|
- #include "Audio.h"
- #include "arm_math.h"
-
-
-
-
- static arm_cfft_radix4_instance_q15 fft_inst;
-
- void AudioAnalyzeFFT256::init(void)
- {
- // TODO: replace this with static const version
- arm_cfft_radix4_init_q15(&fft_inst, 256, 0, 1);
-
- //for (int i=0; i<2048; i++) {
- //buffer[i] = i * 3;
- //}
- //__disable_irq();
- //ARM_DEMCR |= ARM_DEMCR_TRCENA;
- //ARM_DWT_CTRL |= ARM_DWT_CTRL_CYCCNTENA;
- //uint32_t n = ARM_DWT_CYCCNT;
- //arm_cfft_radix2_q15(&fft_inst, buffer);
- //n = ARM_DWT_CYCCNT - n;
- //__enable_irq();
- //cycles = n;
- //arm_cmplx_mag_q15(buffer, buffer, 512);
-
- // each audio block is 278525 cycles @ 96 MHz
- // 256 point fft2 takes 65408 cycles
- // 256 point fft4 takes 49108 cycles
- // 128 point cmag takes 10999 cycles
- // 1024 point fft2 takes 125948 cycles
- // 1024 point fft4 takes 125840 cycles
- // 512 point cmag takes 43764 cycles
-
- //state = 0;
- //outputflag = false;
- }
-
- static void copy_to_fft_buffer(void *destination, const void *source)
- {
- const int16_t *src = (const int16_t *)source;
- int16_t *dst = (int16_t *)destination;
-
- // TODO: optimize this
- for (int i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
- *dst++ = *src++; // real
- *dst++ = 0; // imaginary
- }
- }
-
- // computes limit((val >> rshift), 2**bits)
- static inline int32_t signed_saturate_rshift(int32_t val, int bits, int rshift) __attribute__((always_inline));
- static inline int32_t signed_saturate_rshift(int32_t val, int bits, int rshift)
- {
- int32_t out;
- asm volatile("ssat %0, %1, %2, asr %3" : "=r" (out) : "I" (bits), "r" (val), "I" (rshift));
- return out;
- }
-
- static void apply_window_to_fft_buffer(void *buffer, const void *window)
- {
- int16_t *buf = (int16_t *)buffer;
- const int16_t *win = (int16_t *)window;;
-
- for (int i=0; i < 256; i++) {
- int32_t val = *buf * *win++;
- //*buf = signed_saturate_rshift(val, 16, 15);
- *buf = val >> 15;
- buf += 2;
- }
-
- }
-
- void AudioAnalyzeFFT256::update(void)
- {
- audio_block_t *block;
-
- block = receiveReadOnly();
- if (!block) return;
- if (!prevblock) {
- prevblock = block;
- return;
- }
- copy_to_fft_buffer(buffer, prevblock->data);
- copy_to_fft_buffer(buffer+256, block->data);
- //window = AudioWindowBlackmanNuttall256;
- //window = NULL;
- if (window) apply_window_to_fft_buffer(buffer, window);
- arm_cfft_radix4_q15(&fft_inst, buffer);
- // TODO: is this averaging correct? G. Heinzel's paper says we're
- // supposed to average the magnitude squared, then do the square
- // root at the end after dividing by naverage.
- arm_cmplx_mag_q15(buffer, buffer, 128);
- if (count == 0) {
- for (int i=0; i < 128; i++) {
- output[i] = buffer[i];
- }
- } else {
- for (int i=0; i < 128; i++) {
- output[i] += buffer[i];
- }
- }
- if (++count == naverage) {
- count = 0;
- for (int i=0; i < 128; i++) {
- output[i] /= naverage;
- }
- outputflag = true;
- }
-
- release(prevblock);
- prevblock = block;
-
- #if 0
- block = receiveReadOnly();
- if (state == 0) {
- //Serial.print("0");
- if (block == NULL) return;
- copy_to_fft_buffer(buffer, block->data);
- state = 1;
- } else if (state == 1) {
- //Serial.print("1");
- if (block == NULL) return;
- copy_to_fft_buffer(buffer+256, block->data);
- arm_cfft_radix4_q15(&fft_inst, buffer);
- state = 2;
- } else {
- //Serial.print("2");
- arm_cmplx_mag_q15(buffer, output, 128);
- outputflag = true;
- if (block == NULL) return;
- copy_to_fft_buffer(buffer, block->data);
- state = 1;
- }
- release(block);
- #endif
- }
-
-
-
- #ifdef ORIGINAL_AUDIOSYNTHWAVEFORM
- /******************************************************************/
- // PAH - add ramp-up and ramp-down to the onset of the wave
- // the length is specified in samples
- void AudioSynthWaveform::set_ramp_length(uint16_t r_length)
- {
- if(r_length < 0) {
- ramp_length = 0;
- return;
- }
- // Don't set the ramp length longer than about 4 milliseconds
- if(r_length > 44*4) {
- ramp_length = 44*4;
- return;
- }
- ramp_length = r_length;
- }
-
- void AudioSynthWaveform::update(void)
- {
- audio_block_t *block;
- uint32_t i, ph, inc, index, scale;
- int32_t val1, val2, val3;
-
- //Serial.println("AudioSynthWaveform::update");
- if (((magnitude > 0) || ramp_down) && (block = allocate()) != NULL) {
- ph = phase;
- inc = phase_increment;
- for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
- index = ph >> 24;
- val1 = wavetable[index];
- val2 = wavetable[index+1];
- scale = (ph >> 8) & 0xFFFF;
- val2 *= scale;
- val1 *= 0xFFFF - scale;
- val3 = (val1 + val2) >> 16;
-
-
- // The value of ramp_up is always initialized to RAMP_LENGTH and then is
- // decremented each time through here until it reaches zero.
- // The value of ramp_up is used to generate a Q15 fraction which varies
- // from [0 - 1), and multiplies this by the current sample
- if(ramp_up) {
- // ramp up to the new magnitude
- // ramp_mag is the Q15 representation of the fraction
- // Since ramp_up can't be zero, this cannot generate +1
- ramp_mag = ((ramp_length-ramp_up)<<15)/ramp_length;
- ramp_up--;
- block->data[i] = (val3 * ((ramp_mag * magnitude)>>15)) >> 15;
-
- } else if(ramp_down) {
- // ramp down to zero from the last magnitude
- // The value of ramp_down is always initialized to RAMP_LENGTH and then is
- // decremented each time through here until it reaches zero.
- // The value of ramp_down is used to generate a Q15 fraction which varies
- // from (1 - 0], and multiplies this by the current sample
- // avoid RAMP_LENGTH/RAMP_LENGTH because Q15 format
- // cannot represent +1
- ramp_mag = ((ramp_down - 1)<<15)/ramp_length;
- ramp_down--;
- block->data[i] = (val3 * ((ramp_mag * last_magnitude)>>15)) >> 15;
- } else {
- block->data[i] = (val3 * magnitude) >> 15;
- }
-
- //Serial.print(block->data[i]);
- //Serial.print(", ");
- //if ((i % 12) == 11) Serial.println();
- ph += inc;
- }
- //Serial.println();
- phase = ph;
- transmit(block);
- release(block);
- } else {
- // is this numerical overflow ok?
- phase += phase_increment * AUDIO_BLOCK_SAMPLES;
- }
- }
- #else
- /******************************************************************/
- // PAH - add ramp-up and ramp-down to the onset of the wave
- // the length is specified in samples
- void AudioSynthWaveform::set_ramp_length(uint16_t r_length)
- {
- if(r_length < 0) {
- ramp_length = 0;
- return;
- }
- // Don't set the ramp length longer than about 4 milliseconds
- if(r_length > 44*4) {
- ramp_length = 44*4;
- return;
- }
- ramp_length = r_length;
- }
-
- boolean AudioSynthWaveform::begin(float t_amp,int t_hi,short type)
- {
- tone_type = type;
- // tone_amp = t_amp;
- amplitude(t_amp);
- tone_freq = t_hi;
- if(t_hi < 1)return false;
- if(t_hi >= AUDIO_SAMPLE_RATE_EXACT/2)return false;
- tone_phase = 0;
- tone_incr = (0x100000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;
- if(0) {
- Serial.print("AudioSynthWaveform.begin(tone_amp = ");
- Serial.print(t_amp);
- Serial.print(", tone_hi = ");
- Serial.print(t_hi);
- Serial.print(", tone_incr = ");
- Serial.print(tone_incr,HEX);
- // Serial.print(", tone_hi = ");
- // Serial.print(t_hi);
- Serial.println(")");
- }
- return(true);
- }
-
-
- void AudioSynthWaveform::update(void)
- {
- audio_block_t *block;
- short *bp;
- // temporary for ramp in sine
- uint32_t ramp_mag;
- // temporaries for TRIANGLE
- uint32_t mag;
- short tmp_amp;
-
-
- if(tone_freq == 0)return;
- // L E F T C H A N N E L O N L Y
- block = allocate();
- if(block) {
- bp = block->data;
- switch(tone_type) {
- case TONE_TYPE_SINE:
- for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
- // The value of ramp_up is always initialized to RAMP_LENGTH and then is
- // decremented each time through here until it reaches zero.
- // The value of ramp_up is used to generate a Q15 fraction which varies
- // from [0 - 1), and multiplies this by the current sample
- if(ramp_up) {
- // ramp up to the new magnitude
- // ramp_mag is the Q15 representation of the fraction
- // Since ramp_up can't be zero, this cannot generate +1
- ramp_mag = ((ramp_length-ramp_up)<<15)/ramp_length;
- ramp_up--;
- // adjust tone_phase to Q15 format and then adjust the result
- // of the multiplication
- *bp = (short)((arm_sin_q15(tone_phase>>17) * tone_amp) >> 15);
- *bp++ = (*bp * ramp_mag)>>15;
- }
- else if(ramp_down) {
- // ramp down to zero from the last magnitude
- // The value of ramp_down is always initialized to RAMP_LENGTH and then is
- // decremented each time through here until it reaches zero.
- // The value of ramp_down is used to generate a Q15 fraction which varies
- // from (1 - 0], and multiplies this by the current sample
- // avoid RAMP_LENGTH/RAMP_LENGTH because Q15 format
- // cannot represent +1
- ramp_mag = ((ramp_down - 1)<<15)/ramp_length;
- ramp_down--;
- // adjust tone_phase to Q15 format and then adjust the result
- // of the multiplication
- *bp = (short)((arm_sin_q15(tone_phase>>17) * last_tone_amp) >> 15);
- *bp++ = (*bp * ramp_mag)>>15;
- } else {
- // adjust tone_phase to Q15 format and then adjust the result
- // of the multiplication
- *bp++ = (short)((arm_sin_q15(tone_phase>>17) * tone_amp) >> 15);
- }
-
- // phase and incr are both unsigned 32-bit fractions
- tone_phase += tone_incr;
- }
- break;
-
- case TONE_TYPE_SQUARE:
- for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
- if(tone_phase & 0x80000000)*bp++ = -tone_amp;
- else *bp++ = tone_amp;
- // phase and incr are both unsigned 32-bit fractions
- tone_phase += tone_incr;
- }
- break;
-
- case TONE_TYPE_SAWTOOTH:
- for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
- *bp = ((short)(tone_phase>>16)*tone_amp) >> 15;
- bp++;
- // phase and incr are both unsigned 32-bit fractions
- tone_phase += tone_incr;
- }
- break;
-
- case TONE_TYPE_TRIANGLE:
- for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
- if(tone_phase & 0x80000000) {
- // negative half-cycle
- tmp_amp = -tone_amp;
- }
- else {
- // positive half-cycle
- tmp_amp = tone_amp;
- }
- mag = tone_phase << 2;
- // Determine which quadrant
- if(tone_phase & 0x40000000) {
- // negate the magnitude
- mag = ~mag + 1;
- }
- *bp++ = ((short)(mag>>17)*tmp_amp) >> 15;
- tone_phase += tone_incr;
- }
- break;
- }
- // send the samples to the left channel
- transmit(block,0);
- release(block);
- }
- }
-
-
- #endif
-
-
-
-
-
-
-
-
- #if 0
- void AudioSineWaveMod::frequency(float f)
- {
- if (f > AUDIO_SAMPLE_RATE_EXACT / 2 || f < 0.0) return;
- phase_increment = (f / AUDIO_SAMPLE_RATE_EXACT) * 4294967296.0f;
- }
-
- void AudioSineWaveMod::update(void)
- {
- audio_block_t *block, *modinput;
- uint32_t i, ph, inc, index, scale;
- int32_t val1, val2;
-
- //Serial.println("AudioSineWave::update");
- modinput = receiveReadOnly();
- ph = phase;
- inc = phase_increment;
- block = allocate();
- if (!block) {
- // unable to allocate memory, so we'll send nothing
- if (modinput) {
- // but if we got modulation data, update the phase
- for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
- ph += inc + modinput->data[i] * modulation_factor;
- }
- release(modinput);
- } else {
- ph += phase_increment * AUDIO_BLOCK_SAMPLES;
- }
- phase = ph;
- return;
- }
- if (modinput) {
- for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
- index = ph >> 24;
- val1 = sine_table[index];
- val2 = sine_table[index+1];
- scale = (ph >> 8) & 0xFFFF;
- val2 *= scale;
- val1 *= 0xFFFF - scale;
- block->data[i] = (val1 + val2) >> 16;
- //Serial.print(block->data[i]);
- //Serial.print(", ");
- //if ((i % 12) == 11) Serial.println();
- ph += inc + modinput->data[i] * modulation_factor;
- }
- release(modinput);
- } else {
- ph = phase;
- inc = phase_increment;
- for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
- index = ph >> 24;
- val1 = sine_table[index];
- val2 = sine_table[index+1];
- scale = (ph >> 8) & 0xFFFF;
- val2 *= scale;
- val1 *= 0xFFFF - scale;
- block->data[i] = (val1 + val2) >> 16;
- //Serial.print(block->data[i]);
- //Serial.print(", ");
- //if ((i % 12) == 11) Serial.println();
- ph += inc;
- }
- }
- phase = ph;
- transmit(block);
- release(block);
- }
- #endif
-
-
-
-
-
-
- /******************************************************************/
-
-
- void AudioPrint::update(void)
- {
- audio_block_t *block;
- uint32_t i;
-
- Serial.println("AudioPrint::update");
- Serial.println(name);
- block = receiveReadOnly();
- if (block) {
- for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
- Serial.print(block->data[i]);
- Serial.print(", ");
- if ((i % 12) == 11) Serial.println();
- }
- Serial.println();
- release(block);
- }
- }
-
-
-
- /******************************************************************/
-
-
- #define STATE_DIRECT_8BIT_MONO 0 // playing mono at native sample rate
- #define STATE_DIRECT_8BIT_STEREO 1 // playing stereo at native sample rate
- #define STATE_DIRECT_16BIT_MONO 2 // playing mono at native sample rate
- #define STATE_DIRECT_16BIT_STEREO 3 // playing stereo at native sample rate
- #define STATE_CONVERT_8BIT_MONO 4 // playing mono, converting sample rate
- #define STATE_CONVERT_8BIT_STEREO 5 // playing stereo, converting sample rate
- #define STATE_CONVERT_16BIT_MONO 6 // playing mono, converting sample rate
- #define STATE_CONVERT_16BIT_STEREO 7 // playing stereo, converting sample rate
- #define STATE_PARSE1 8 // looking for 20 byte ID header
- #define STATE_PARSE2 9 // looking for 16 byte format header
- #define STATE_PARSE3 10 // looking for 8 byte data header
- #define STATE_PARSE4 11 // ignoring unknown chunk
- #define STATE_STOP 12
-
- void AudioPlaySDcardWAV::begin(void)
- {
- state = STATE_STOP;
- state_play = STATE_STOP;
- data_length = 0;
- if (block_left) {
- release(block_left);
- block_left = NULL;
- }
- if (block_right) {
- release(block_right);
- block_right = NULL;
- }
- }
-
-
- bool AudioPlaySDcardWAV::play(const char *filename)
- {
- stop();
- wavfile = SD.open(filename);
- if (!wavfile) return false;
- buffer_remaining = 0;
- state_play = STATE_STOP;
- data_length = 0;
- state = STATE_PARSE1;
- return true;
- }
-
- void AudioPlaySDcardWAV::stop(void)
- {
- __disable_irq();
- if (state != STATE_STOP) {
- audio_block_t *b1 = block_left;
- block_left = NULL;
- audio_block_t *b2 = block_right;
- block_right = NULL;
- state = STATE_STOP;
- __enable_irq();
- if (b1) release(b1);
- if (b2) release(b2);
- wavfile.close();
- } else {
- __enable_irq();
- }
- }
-
- bool AudioPlaySDcardWAV::start(void)
- {
- __disable_irq();
- if (state == STATE_STOP) {
- if (state_play == STATE_STOP) {
- __enable_irq();
- return false;
- }
- state = state_play;
- }
- __enable_irq();
- return true;
- }
-
-
- void AudioPlaySDcardWAV::update(void)
- {
- // only update if we're playing
- if (state == STATE_STOP) return;
-
- // allocate the audio blocks to transmit
- block_left = allocate();
- if (block_left == NULL) return;
- if (state < 8 && (state & 1) == 1) {
- // if we're playing stereo, allocate another
- // block for the right channel output
- block_right = allocate();
- if (block_right == NULL) {
- release(block_left);
- return;
- }
- } else {
- // if we're playing mono or just parsing
- // the WAV file header, no right-side block
- block_right = NULL;
- }
- block_offset = 0;
-
- //Serial.println("update");
-
- // is there buffered data?
- if (buffer_remaining > 0) {
- // we have buffered data
- if (consume()) return; // it was enough to transmit audio
- }
-
- // we only get to this point when buffer[512] is empty
- if (state != STATE_STOP && wavfile.available()) {
- // we can read more data from the file...
- buffer_remaining = wavfile.read(buffer, 512);
- if (consume()) {
- // good, it resulted in audio transmit
- return;
- } else {
- // not good, no audio was transmitted
- buffer_remaining = 0;
- if (block_left) {
- release(block_left);
- block_left = NULL;
- }
- if (block_right) {
- release(block_right);
- block_right = NULL;
- }
- // if we're still playing, well, there's going to
- // be a gap in output, but we can't keep burning
- // time trying to read more data. Hopefully things
- // will go better next time?
- if (state != STATE_STOP) return;
- }
- }
- // end of file reached or other reason to stop
- wavfile.close();
- if (block_left) {
- release(block_left);
- block_left = NULL;
- }
- if (block_right) {
- release(block_right);
- block_right = NULL;
- }
- state_play = STATE_STOP;
- state = STATE_STOP;
- }
-
-
- // https://ccrma.stanford.edu/courses/422/projects/WaveFormat/
-
- // Consume already buffered data. Returns true if audio transmitted.
- bool AudioPlaySDcardWAV::consume(void)
- {
- uint32_t len, size;
- uint8_t lsb, msb;
- const uint8_t *p;
-
- size = buffer_remaining;
- p = buffer + 512 - size;
- start:
- if (size == 0) return false;
- //Serial.print("AudioPlaySDcardWAV write, size = ");
- //Serial.print(size);
- //Serial.print(", data_length = ");
- //Serial.print(data_length);
- //Serial.print(", state = ");
- //Serial.println(state);
- switch (state) {
- // parse wav file header, is this really a .wav file?
- case STATE_PARSE1:
- len = 20 - data_length;
- if (size < len) len = size;
- memcpy((uint8_t *)header + data_length, p, len);
- data_length += len;
- if (data_length < 20) return false;
- // parse the header...
- if (header[0] == 0x46464952 && header[2] == 0x45564157
- && header[3] == 0x20746D66 && header[4] == 16) {
- //Serial.println("header ok");
- state = STATE_PARSE2;
- p += len;
- size -= len;
- data_length = 0;
- goto start;
- }
- //Serial.println("unknown WAV header");
- break;
-
- // check & extract key audio parameters
- case STATE_PARSE2:
- len = 16 - data_length;
- if (size < len) len = size;
- memcpy((uint8_t *)header + data_length, p, len);
- data_length += len;
- if (data_length < 16) return false;
- if (parse_format()) {
- //Serial.println("audio format ok");
- p += len;
- size -= len;
- data_length = 0;
- state = STATE_PARSE3;
- goto start;
- }
- //Serial.println("unknown audio format");
- break;
-
- // find the data chunk
- case STATE_PARSE3:
- len = 8 - data_length;
- if (size < len) len = size;
- memcpy((uint8_t *)header + data_length, p, len);
- data_length += len;
- if (data_length < 8) return false;
- //Serial.print("chunk id = ");
- //Serial.print(header[0], HEX);
- //Serial.print(", length = ");
- //Serial.println(header[1]);
- p += len;
- size -= len;
- data_length = header[1];
- if (header[0] == 0x61746164) {
- //Serial.println("found data chunk");
- // TODO: verify offset in file is an even number
- // as required by WAV format. abort if odd. Code
- // below will depend upon this and fail if not even.
- leftover_bytes = 0;
- state = state_play;
- if (state & 1) {
- // if we're going to start stereo
- // better allocate another output block
- block_right = allocate();
- if (!block_right) return false;
- }
- } else {
- state = STATE_PARSE4;
- }
- goto start;
-
- // ignore any extra unknown chunks (title & artist info)
- case STATE_PARSE4:
- if (size < data_length) {
- data_length -= size;
- return false;
- }
- p += data_length;
- size -= data_length;
- data_length = 0;
- state = STATE_PARSE3;
- goto start;
-
- // playing mono at native sample rate
- case STATE_DIRECT_8BIT_MONO:
- return false;
-
- // playing stereo at native sample rate
- case STATE_DIRECT_8BIT_STEREO:
- return false;
-
- // playing mono at native sample rate
- case STATE_DIRECT_16BIT_MONO:
- if (size > data_length) size = data_length;
- data_length -= size;
- while (1) {
- lsb = *p++;
- msb = *p++;
- size -= 2;
- block_left->data[block_offset++] = (msb << 8) | lsb;
- if (block_offset >= AUDIO_BLOCK_SAMPLES) {
- transmit(block_left, 0);
- transmit(block_left, 1);
- //Serial1.print('%');
- //delayMicroseconds(90);
- release(block_left);
- block_left = NULL;
- data_length += size;
- buffer_remaining = size;
- if (block_right) release(block_right);
- return true;
- }
- if (size == 0) {
- if (data_length == 0) break;
- return false;
- }
- }
- // end of file reached
- if (block_offset > 0) {
- // TODO: fill remainder of last block with zero and transmit
- }
- state = STATE_STOP;
- return false;
-
- // playing stereo at native sample rate
- case STATE_DIRECT_16BIT_STEREO:
- if (size > data_length) size = data_length;
- data_length -= size;
- if (leftover_bytes) {
- block_left->data[block_offset] = header[0];
- goto right16;
- }
- while (1) {
- lsb = *p++;
- msb = *p++;
- size -= 2;
- if (size == 0) {
- if (data_length == 0) break;
- header[0] = (msb << 8) | lsb;
- leftover_bytes = 2;
- return false;
- }
- block_left->data[block_offset] = (msb << 8) | lsb;
- right16:
- lsb = *p++;
- msb = *p++;
- size -= 2;
- block_right->data[block_offset++] = (msb << 8) | lsb;
- if (block_offset >= AUDIO_BLOCK_SAMPLES) {
- transmit(block_left, 0);
- release(block_left);
- block_left = NULL;
- transmit(block_right, 1);
- release(block_right);
- block_right = NULL;
- data_length += size;
- buffer_remaining = size;
- return true;
- }
- if (size == 0) {
- if (data_length == 0) break;
- leftover_bytes = 0;
- return false;
- }
- }
- // end of file reached
- if (block_offset > 0) {
- // TODO: fill remainder of last block with zero and transmit
- }
- state = STATE_STOP;
- return false;
-
- // playing mono, converting sample rate
- case STATE_CONVERT_8BIT_MONO :
- return false;
-
- // playing stereo, converting sample rate
- case STATE_CONVERT_8BIT_STEREO:
- return false;
-
- // playing mono, converting sample rate
- case STATE_CONVERT_16BIT_MONO:
- return false;
-
- // playing stereo, converting sample rate
- case STATE_CONVERT_16BIT_STEREO:
- return false;
-
- // ignore any extra data after playing
- // or anything following any error
- case STATE_STOP:
- return false;
-
- // this is not supposed to happen!
- //default:
- //Serial.println("AudioPlaySDcardWAV, unknown state");
- }
- state_play = STATE_STOP;
- state = STATE_STOP;
- return false;
- }
-
-
- /*
- 00000000 52494646 66EA6903 57415645 666D7420 RIFFf.i.WAVEfmt
- 00000010 10000000 01000200 44AC0000 10B10200 ........D.......
- 00000020 04001000 4C495354 3A000000 494E464F ....LIST:...INFO
- 00000030 494E414D 14000000 49205761 6E742054 INAM....I Want T
- 00000040 6F20436F 6D65204F 76657200 49415254 o Come Over.IART
- 00000050 12000000 4D656C69 73736120 45746865 ....Melissa Ethe
- 00000060 72696467 65006461 746100EA 69030100 ridge.data..i...
- 00000070 FEFF0300 FCFF0400 FDFF0200 0000FEFF ................
- 00000080 0300FDFF 0200FFFF 00000100 FEFF0300 ................
- 00000090 FDFF0300 FDFF0200 FFFF0100 0000FFFF ................
- */
-
-
-
-
-
- // SD library on Teensy3 at 96 MHz
- // 256 byte chunks, speed is 443272 bytes/sec
- // 512 byte chunks, speed is 468023 bytes/sec
-
-
-
-
-
- bool AudioPlaySDcardWAV::parse_format(void)
- {
- uint8_t num = 0;
- uint16_t format;
- uint16_t channels;
- uint32_t rate;
- uint16_t bits;
-
- format = header[0];
- //Serial.print(" format = ");
- //Serial.println(format);
- if (format != 1) return false;
-
- channels = header[0] >> 16;
- //Serial.print(" channels = ");
- //Serial.println(channels);
- if (channels == 1) {
- } else if (channels == 2) {
- num = 1;
- } else {
- return false;
- }
-
- bits = header[3] >> 16;
- //Serial.print(" bits = ");
- //Serial.println(bits);
- if (bits == 8) {
- } else if (bits == 16) {
- num |= 2;
- } else {
- return false;
- }
-
- rate = header[1];
- //Serial.print(" rate = ");
- //Serial.println(rate);
- if (rate == AUDIO_SAMPLE_RATE) {
- } else if (rate >= 8000 && rate <= 48000) {
- num |= 4;
- } else {
- return false;
- }
- // we're not checking the byte rate and block align fields
- // if they're not the expected values, all we could do is
- // return false. Do any real wav files have unexpected
- // values in these other fields?
- state_play = num;
- return true;
- }
-
-
-
- /******************************************************************/
-
-
- void AudioPlaySDcardRAW::begin(void)
- {
- playing = false;
- if (block) {
- release(block);
- block = NULL;
- }
- }
-
-
- bool AudioPlaySDcardRAW::play(const char *filename)
- {
- stop();
- rawfile = SD.open(filename);
- if (!rawfile) {
- Serial.println("unable to open file");
- return false;
- }
- Serial.println("able to open file");
- playing = true;
- return true;
- }
-
- void AudioPlaySDcardRAW::stop(void)
- {
- __disable_irq();
- if (playing) {
- playing = false;
- __enable_irq();
- rawfile.close();
- } else {
- __enable_irq();
- }
- }
-
-
- void AudioPlaySDcardRAW::update(void)
- {
- unsigned int i, n;
-
- // only update if we're playing
- if (!playing) return;
-
- // allocate the audio blocks to transmit
- block = allocate();
- if (block == NULL) return;
-
- if (rawfile.available()) {
- // we can read more data from the file...
- n = rawfile.read(block->data, AUDIO_BLOCK_SAMPLES*2);
- for (i=n/2; i < AUDIO_BLOCK_SAMPLES; i++) {
- block->data[i] = 0;
- }
- transmit(block);
- release(block);
- } else {
- rawfile.close();
- playing = false;
- }
- }
-
-
- /******************************************************************/
-
-
-
- void AudioPlayMemory::play(const unsigned int *data)
- {
- uint32_t format;
-
- playing = 0;
- prior = 0;
- format = *data++;
- next = data;
- length = format & 0xFFFFFF;
- playing = format >> 24;
- }
-
- void AudioPlayMemory::stop(void)
- {
- playing = 0;
- }
-
- extern "C" {
- extern const int16_t ulaw_decode_table[256];
- };
-
- void AudioPlayMemory::update(void)
- {
- audio_block_t *block;
- const unsigned int *in;
- int16_t *out;
- uint32_t tmp32, consumed;
- int16_t s0, s1, s2, s3, s4;
- int i;
-
- if (!playing) return;
- block = allocate();
- if (block == NULL) return;
-
- //Serial.write('.');
-
- out = block->data;
- in = next;
- s0 = prior;
-
- switch (playing) {
- case 0x01: // u-law encoded, 44100 Hz
- for (i=0; i < AUDIO_BLOCK_SAMPLES; i += 4) {
- tmp32 = *in++;
- *out++ = ulaw_decode_table[(tmp32 >> 0) & 255];
- *out++ = ulaw_decode_table[(tmp32 >> 8) & 255];
- *out++ = ulaw_decode_table[(tmp32 >> 16) & 255];
- *out++ = ulaw_decode_table[(tmp32 >> 24) & 255];
- }
- consumed = 128;
- break;
-
- case 0x81: // 16 bit PCM, 44100 Hz
- for (i=0; i < AUDIO_BLOCK_SAMPLES; i += 2) {
- tmp32 = *in++;
- *out++ = (int16_t)(tmp32 & 65535);
- *out++ = (int16_t)(tmp32 >> 16);
- }
- consumed = 128;
- break;
-
- case 0x02: // u-law encoded, 22050 Hz
- for (i=0; i < AUDIO_BLOCK_SAMPLES; i += 8) {
- tmp32 = *in++;
- s1 = ulaw_decode_table[(tmp32 >> 0) & 255];
- s2 = ulaw_decode_table[(tmp32 >> 8) & 255];
- s3 = ulaw_decode_table[(tmp32 >> 16) & 255];
- s4 = ulaw_decode_table[(tmp32 >> 24) & 255];
- *out++ = (s0 + s1) >> 1;
- *out++ = s1;
- *out++ = (s1 + s2) >> 1;
- *out++ = s2;
- *out++ = (s2 + s3) >> 1;
- *out++ = s3;
- *out++ = (s3 + s4) >> 1;
- *out++ = s4;
- s0 = s4;
- }
- consumed = 64;
- break;
-
- case 0x82: // 16 bits PCM, 22050 Hz
- for (i=0; i < AUDIO_BLOCK_SAMPLES; i += 4) {
- tmp32 = *in++;
- s1 = (int16_t)(tmp32 & 65535);
- s2 = (int16_t)(tmp32 >> 16);
- *out++ = (s0 + s1) >> 1;
- *out++ = s1;
- *out++ = (s1 + s2) >> 1;
- *out++ = s2;
- s0 = s2;
- }
- consumed = 64;
- break;
-
- case 0x03: // u-law encoded, 11025 Hz
- for (i=0; i < AUDIO_BLOCK_SAMPLES; i += 16) {
- tmp32 = *in++;
- s1 = ulaw_decode_table[(tmp32 >> 0) & 255];
- s2 = ulaw_decode_table[(tmp32 >> 8) & 255];
- s3 = ulaw_decode_table[(tmp32 >> 16) & 255];
- s4 = ulaw_decode_table[(tmp32 >> 24) & 255];
- *out++ = (s0 * 3 + s1) >> 2;
- *out++ = (s0 + s1) >> 1;
- *out++ = (s0 + s1 * 3) >> 2;
- *out++ = s1;
- *out++ = (s1 * 3 + s2) >> 2;
- *out++ = (s1 + s2) >> 1;
- *out++ = (s1 + s2 * 3) >> 2;
- *out++ = s2;
- *out++ = (s2 * 3 + s3) >> 2;
- *out++ = (s2 + s3) >> 1;
- *out++ = (s2 + s3 * 3) >> 2;
- *out++ = s3;
- *out++ = (s3 * 3 + s4) >> 2;
- *out++ = (s3 + s4) >> 1;
- *out++ = (s3 + s4 * 3) >> 2;
- *out++ = s4;
- s0 = s4;
- }
- consumed = 32;
- break;
-
- case 0x83: // 16 bit PCM, 11025 Hz
- for (i=0; i < AUDIO_BLOCK_SAMPLES; i += 8) {
- tmp32 = *in++;
- s1 = (int16_t)(tmp32 & 65535);
- s2 = (int16_t)(tmp32 >> 16);
- *out++ = (s0 * 3 + s1) >> 2;
- *out++ = (s0 + s1) >> 1;
- *out++ = (s0 + s1 * 3) >> 2;
- *out++ = s1;
- *out++ = (s1 * 3 + s2) >> 2;
- *out++ = (s1 + s2) >> 1;
- *out++ = (s1 + s2 * 3) >> 2;
- *out++ = s2;
- s0 = s2;
- }
- consumed = 32;
- break;
-
- default:
- release(block);
- playing = 0;
- return;
- }
- prior = s0;
- next = in;
- if (length > consumed) {
- length -= consumed;
- } else {
- playing = 0;
- }
- transmit(block);
- release(block);
- }
-
-
-
-
-
-
-
-
-
-
- /******************************************************************/
-
-
-
-
- // computes ((a[31:0] * b[15:0]) >> 16)
- static inline int32_t signed_multiply_32x16b(int32_t a, uint32_t b) __attribute__((always_inline));
- static inline int32_t signed_multiply_32x16b(int32_t a, uint32_t b)
- {
- int32_t out;
- asm volatile("smulwb %0, %1, %2" : "=r" (out) : "r" (a), "r" (b));
- return out;
- }
-
- // computes ((a[31:0] * b[31:16]) >> 16)
- static inline int32_t signed_multiply_32x16t(int32_t a, uint32_t b) __attribute__((always_inline));
- static inline int32_t signed_multiply_32x16t(int32_t a, uint32_t b)
- {
- int32_t out;
- asm volatile("smulwt %0, %1, %2" : "=r" (out) : "r" (a), "r" (b));
- return out;
- }
-
- // computes (((int64_t)a[31:0] * (int64_t)b[31:0]) >> 32)
- static inline int32_t multiply_32x32_rshift32(int32_t a, int32_t b) __attribute__((always_inline));
- static inline int32_t multiply_32x32_rshift32(int32_t a, int32_t b)
- {
- int32_t out;
- asm volatile("smmul %0, %1, %2" : "=r" (out) : "r" (a), "r" (b));
- return out;
- }
-
- // computes (((int64_t)a[31:0] * (int64_t)b[31:0] + 0x8000000) >> 32)
- static inline int32_t multiply_32x32_rshift32_rounded(int32_t a, int32_t b) __attribute__((always_inline));
- static inline int32_t multiply_32x32_rshift32_rounded(int32_t a, int32_t b)
- {
- int32_t out;
- asm volatile("smmulr %0, %1, %2" : "=r" (out) : "r" (a), "r" (b));
- return out;
- }
-
- // computes sum + (((int64_t)a[31:0] * (int64_t)b[31:0] + 0x8000000) >> 32)
- static inline int32_t multiply_accumulate_32x32_rshift32_rounded(int32_t sum, int32_t a, int32_t b) __attribute__((always_inline));
- static inline int32_t multiply_accumulate_32x32_rshift32_rounded(int32_t sum, int32_t a, int32_t b)
- {
- int32_t out;
- asm volatile("smmlar %0, %2, %3, %1" : "=r" (out) : "r" (sum), "r" (a), "r" (b));
- return out;
- }
-
- // computes sum - (((int64_t)a[31:0] * (int64_t)b[31:0] + 0x8000000) >> 32)
- static inline int32_t multiply_subtract_32x32_rshift32_rounded(int32_t sum, int32_t a, int32_t b) __attribute__((always_inline));
- static inline int32_t multiply_subtract_32x32_rshift32_rounded(int32_t sum, int32_t a, int32_t b)
- {
- int32_t out;
- asm volatile("smmlsr %0, %2, %3, %1" : "=r" (out) : "r" (sum), "r" (a), "r" (b));
- return out;
- }
-
- // computes ((a[15:0] << 16) | b[15:0])
- static inline uint32_t pack_16x16(int32_t a, int32_t b) __attribute__((always_inline));
- static inline uint32_t pack_16x16(int32_t a, int32_t b)
- {
- int32_t out;
- asm volatile("pkhbt %0, %1, %2, lsl #16" : "=r" (out) : "r" (b), "r" (a));
- return out;
- }
-
- // computes (((a[31:16] + b[31:16]) << 16) | (a[15:0 + b[15:0]))
- static inline uint32_t signed_add_16_and_16(uint32_t a, uint32_t b) __attribute__((always_inline));
- static inline uint32_t signed_add_16_and_16(uint32_t a, uint32_t b)
- {
- int32_t out;
- asm volatile("qadd16 %0, %1, %2" : "=r" (out) : "r" (a), "r" (b));
- return out;
- }
-
- // computes (sum + ((a[31:0] * b[15:0]) >> 16))
- static inline int32_t signed_multiply_accumulate_32x16b(int32_t sum, int32_t a, uint32_t b)
- {
- int32_t out;
- asm volatile("smlawb %0, %2, %3, %1" : "=r" (out) : "r" (sum), "r" (a), "r" (b));
- return out;
- }
-
- // computes (sum + ((a[31:0] * b[31:16]) >> 16))
- static inline int32_t signed_multiply_accumulate_32x16t(int32_t sum, int32_t a, uint32_t b)
- {
- int32_t out;
- asm volatile("smlawt %0, %2, %3, %1" : "=r" (out) : "r" (sum), "r" (a), "r" (b));
- return out;
- }
-
- // computes logical and, forces compiler to allocate register and use single cycle instruction
- static inline uint32_t logical_and(uint32_t a, uint32_t b)
- {
- asm volatile("and %0, %1" : "+r" (a) : "r" (b));
- return a;
- }
-
-
-
-
- void applyGain(int16_t *data, int32_t mult)
- {
- uint32_t *p = (uint32_t *)data;
- const uint32_t *end = (uint32_t *)(data + AUDIO_BLOCK_SAMPLES);
-
- do {
- uint32_t tmp32 = *p; // read 2 samples from *data
- int32_t val1 = signed_multiply_32x16b(mult, tmp32);
- int32_t val2 = signed_multiply_32x16t(mult, tmp32);
- val1 = signed_saturate_rshift(val1, 16, 0);
- val2 = signed_saturate_rshift(val2, 16, 0);
- *p++ = pack_16x16(val2, val1);
- } while (p < end);
- }
-
- // page 133
-
- void applyGainThenAdd(int16_t *data, const int16_t *in, int32_t mult)
- {
- uint32_t *dst = (uint32_t *)data;
- const uint32_t *src = (uint32_t *)in;
- const uint32_t *end = (uint32_t *)(data + AUDIO_BLOCK_SAMPLES);
-
- if (mult == 65536) {
- do {
- uint32_t tmp32 = *dst;
- *dst++ = signed_add_16_and_16(tmp32, *src++);
- tmp32 = *dst;
- *dst++ = signed_add_16_and_16(tmp32, *src++);
- } while (dst < end);
- } else {
- do {
- uint32_t tmp32 = *src++; // read 2 samples from *data
- int32_t val1 = signed_multiply_32x16b(mult, tmp32);
- int32_t val2 = signed_multiply_32x16t(mult, tmp32);
- val1 = signed_saturate_rshift(val1, 16, 0);
- val2 = signed_saturate_rshift(val2, 16, 0);
- tmp32 = pack_16x16(val2, val1);
- uint32_t tmp32b = *dst;
- *dst++ = signed_add_16_and_16(tmp32, tmp32b);
- } while (dst < end);
- }
- }
-
-
- void AudioMixer4::update(void)
- {
- audio_block_t *in, *out=NULL;
- unsigned int channel;
-
- for (channel=0; channel < 4; channel++) {
- if (!out) {
- out = receiveWritable(channel);
- if (out) {
- int32_t mult = multiplier[channel];
- if (mult != 65536) applyGain(out->data, mult);
- }
- } else {
- in = receiveReadOnly(channel);
- if (in) {
- applyGainThenAdd(out->data, in->data, multiplier[channel]);
- release(in);
- }
- }
- }
- if (out) {
- transmit(out);
- release(out);
- }
- }
-
-
- /******************************************************************/
-
-
-
-
-
- void AudioFilterBiquad::update(void)
- {
- audio_block_t *block;
- int32_t a0, a1, a2, b1, b2, sum;
- uint32_t in2, out2, aprev, bprev, flag;
- uint32_t *data, *end;
- int32_t *state;
-
- block = receiveWritable();
- if (!block) return;
- data = (uint32_t *)(block->data);
- end = data + AUDIO_BLOCK_SAMPLES/2;
- state = (int32_t *)definition;
- do {
- a0 = *state++;
- a1 = *state++;
- a2 = *state++;
- b1 = *state++;
- b2 = *state++;
- aprev = *state++;
- bprev = *state++;
- sum = *state & 0x3FFF;
- do {
- in2 = *data;
- sum = signed_multiply_accumulate_32x16b(sum, a0, in2);
- sum = signed_multiply_accumulate_32x16t(sum, a1, aprev);
- sum = signed_multiply_accumulate_32x16b(sum, a2, aprev);
- sum = signed_multiply_accumulate_32x16t(sum, b1, bprev);
- sum = signed_multiply_accumulate_32x16b(sum, b2, bprev);
- out2 = (uint32_t)sum >> 14;
- sum &= 0x3FFF;
- sum = signed_multiply_accumulate_32x16t(sum, a0, in2);
- sum = signed_multiply_accumulate_32x16b(sum, a1, in2);
- sum = signed_multiply_accumulate_32x16t(sum, a2, aprev);
- sum = signed_multiply_accumulate_32x16b(sum, b1, out2);
- sum = signed_multiply_accumulate_32x16t(sum, b2, bprev);
- aprev = in2;
- bprev = pack_16x16(sum >> 14, out2);
- sum &= 0x3FFF;
- aprev = in2;
- *data++ = bprev;
- } while (data < end);
- flag = *state & 0x80000000;
- *state++ = sum | flag;
- *(state-2) = bprev;
- *(state-3) = aprev;
- } while (flag);
- transmit(block);
- release(block);
- }
-
- void AudioFilterBiquad::updateCoefs(int *source, bool doReset)
- {
- int32_t *dest=(int32_t *)definition;
- int32_t *src=(int32_t *)source;
- __disable_irq();
- for(uint8_t index=0;index<5;index++)
- {
- *dest++=*src++;
- }
- if(doReset)
- {
- *dest++=0;
- *dest++=0;
- *dest++=0;
- }
- __enable_irq();
- }
-
- void AudioFilterBiquad::updateCoefs(int *source)
- {
- updateCoefs(source,false);
- }
-
- /******************************************************************/
-
-
- extern "C" {
- extern const int16_t fader_table[256];
- };
-
-
- void AudioEffectFade::update(void)
- {
- audio_block_t *block;
- uint32_t i, pos, inc, index, scale;
- int32_t val1, val2, val, sample;
- uint8_t dir;
-
- pos = position;
- if (pos == 0) {
- // output is silent
- block = receiveReadOnly();
- if (block) release(block);
- return;
- } else if (pos == 0xFFFFFFFF) {
- // output is 100%
- block = receiveReadOnly();
- if (!block) return;
- transmit(block);
- release(block);
- return;
- }
- block = receiveWritable();
- if (!block) return;
- inc = rate;
- dir = direction;
- for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
- index = pos >> 24;
- val1 = fader_table[index];
- val2 = fader_table[index+1];
- scale = (pos >> 8) & 0xFFFF;
- val2 *= scale;
- val1 *= 0x10000 - scale;
- val = (val1 + val2) >> 16;
- sample = block->data[i];
- sample = (sample * val) >> 15;
- block->data[i] = sample;
- if (dir > 0) {
- // output is increasing
- if (inc < 0xFFFFFFFF - pos) pos += inc;
- else pos = 0xFFFFFFFF;
- } else {
- // output is decreasing
- if (inc < pos) pos -= inc;
- else pos = 0;
- }
- }
- position = pos;
- transmit(block);
- release(block);
- }
-
- void AudioEffectFade::fadeBegin(uint32_t newrate, uint8_t dir)
- {
- __disable_irq();
- uint32_t pos = position;
- if (pos == 0) position = 1;
- else if (pos == 0xFFFFFFFF) position = 0xFFFFFFFE;
- rate = newrate;
- direction = dir;
- __enable_irq();
- }
-
-
-
- /******************************************************************/
-
-
- static inline int32_t multiply_32x32_rshift30(int32_t a, int32_t b) __attribute__((always_inline));
- static inline int32_t multiply_32x32_rshift30(int32_t a, int32_t b)
- {
- return ((int64_t)a * (int64_t)b) >> 30;
- }
-
- //#define TONE_DETECT_FAST
-
- void AudioAnalyzeToneDetect::update(void)
- {
- audio_block_t *block;
- int32_t q0, q1, q2, coef;
- const int16_t *p, *end;
- uint16_t n;
-
- block = receiveReadOnly();
- if (!block) return;
- if (!enabled) {
- release(block);
- return;
- }
- p = block->data;
- end = p + AUDIO_BLOCK_SAMPLES;
- n = count;
- coef = coefficient;
- q1 = s1;
- q2 = s2;
- do {
- // the Goertzel algorithm is kinda magical ;-)
- #ifdef TONE_DETECT_FAST
- q0 = (*p++) + (multiply_32x32_rshift32_rounded(coef, q1) << 2) - q2;
- #else
- q0 = (*p++) + multiply_32x32_rshift30(coef, q1) - q2;
- // TODO: is this only 1 cycle slower? if so, always use it
- #endif
- q2 = q1;
- q1 = q0;
- if (--n == 0) {
- out1 = q1;
- out2 = q2;
- q1 = 0; // TODO: does clearing these help or hinder?
- q2 = 0;
- new_output = true;
- n = length;
- }
- } while (p < end);
- count = n;
- s1 = q1;
- s2 = q2;
- release(block);
- }
-
- void AudioAnalyzeToneDetect::set_params(int32_t coef, uint16_t cycles, uint16_t len)
- {
- __disable_irq();
- coefficient = coef;
- ncycles = cycles;
- length = len;
- count = len;
- s1 = 0;
- s2 = 0;
- enabled = true;
- __enable_irq();
- //Serial.printf("Tone: coef=%d, ncycles=%d, length=%d\n", coefficient, ncycles, length);
- }
-
- float AudioAnalyzeToneDetect::read(void)
- {
- int32_t coef, q1, q2, power;
- uint16_t len;
-
- __disable_irq();
- coef = coefficient;
- q1 = out1;
- q2 = out2;
- len = length;
- __enable_irq();
- #ifdef TONE_DETECT_FAST
- power = multiply_32x32_rshift32_rounded(q2, q2);
- power = multiply_accumulate_32x32_rshift32_rounded(power, q1, q1);
- power = multiply_subtract_32x32_rshift32_rounded(power,
- multiply_32x32_rshift30(q1, q2), coef);
- power <<= 4;
- #else
- int64_t power64;
- power64 = (int64_t)q2 * (int64_t)q2;
- power64 += (int64_t)q1 * (int64_t)q1;
- power64 -= (((int64_t)q1 * (int64_t)q2) >> 30) * (int64_t)coef;
- power = power64 >> 28;
- #endif
- return sqrtf((float)power) / (float)len;
- }
-
-
- AudioAnalyzeToneDetect::operator bool()
- {
- int32_t coef, q1, q2, power, trigger;
- uint16_t len;
-
- __disable_irq();
- coef = coefficient;
- q1 = out1;
- q2 = out2;
- len = length;
- __enable_irq();
- #ifdef TONE_DETECT_FAST
- power = multiply_32x32_rshift32_rounded(q2, q2);
- power = multiply_accumulate_32x32_rshift32_rounded(power, q1, q1);
- power = multiply_subtract_32x32_rshift32_rounded(power,
- multiply_32x32_rshift30(q1, q2), coef);
- power <<= 4;
- #else
- int64_t power64;
- power64 = (int64_t)q2 * (int64_t)q2;
- power64 += (int64_t)q1 * (int64_t)q1;
- power64 -= (((int64_t)q1 * (int64_t)q2) >> 30) * (int64_t)coef;
- power = power64 >> 28;
- #endif
- trigger = (uint32_t)len * thresh;
- trigger = multiply_32x32_rshift32(trigger, trigger);
-
- Serial.printf("bool: power=%d, trig=%d\n", power, trigger);
- return (power >= trigger);
- }
-
-
-
-
- /******************************************************************/
-
- #include "Wire.h"
-
- #define WM8731_I2C_ADDR 0x1A
- //#define WM8731_I2C_ADDR 0x1B
-
- #define WM8731_REG_LLINEIN 0
- #define WM8731_REG_RLINEIN 1
- #define WM8731_REG_LHEADOUT 2
- #define WM8731_REG_RHEADOUT 3
- #define WM8731_REG_ANALOG 4
- #define WM8731_REG_DIGITAL 5
- #define WM8731_REG_POWERDOWN 6
- #define WM8731_REG_INTERFACE 7
- #define WM8731_REG_SAMPLING 8
- #define WM8731_REG_ACTIVE 9
- #define WM8731_REG_RESET 15
-
- bool AudioControlWM8731::enable(void)
- {
- Wire.begin();
- delay(5);
- //write(WM8731_REG_RESET, 0);
-
- write(WM8731_REG_INTERFACE, 0x02); // I2S, 16 bit, MCLK slave
- write(WM8731_REG_SAMPLING, 0x20); // 256*Fs, 44.1 kHz, MCLK/1
-
- // In order to prevent pops, the DAC should first be soft-muted (DACMU),
- // the output should then be de-selected from the line and headphone output
- // (DACSEL), then the DAC powered down (DACPD).
-
- write(WM8731_REG_DIGITAL, 0x08); // DAC soft mute
- write(WM8731_REG_ANALOG, 0x00); // disable all
-
- write(WM8731_REG_POWERDOWN, 0x00); // codec powerdown
-
- write(WM8731_REG_LHEADOUT, 0x80); // volume off
- write(WM8731_REG_RHEADOUT, 0x80);
-
- delay(100); // how long to power up?
-
- write(WM8731_REG_ACTIVE, 1);
- delay(5);
- write(WM8731_REG_DIGITAL, 0x00); // DAC unmuted
- write(WM8731_REG_ANALOG, 0x10); // DAC selected
-
- return true;
- }
-
-
- bool AudioControlWM8731::write(unsigned int reg, unsigned int val)
- {
- Wire.beginTransmission(WM8731_I2C_ADDR);
- Wire.write((reg << 1) | ((val >> 8) & 1));
- Wire.write(val & 0xFF);
- Wire.endTransmission();
- return true;
- }
-
- bool AudioControlWM8731::volumeInteger(unsigned int n)
- {
- if (n > 127) n = 127;
- //Serial.print("volumeInteger, n = ");
- //Serial.println(n);
- write(WM8731_REG_LHEADOUT, n | 0x180);
- write(WM8731_REG_RHEADOUT, n | 0x80);
- return true;
- }
-
-
-
- /******************************************************************/
-
-
- bool AudioControlWM8731master::enable(void)
- {
- Wire.begin();
- delay(5);
- //write(WM8731_REG_RESET, 0);
-
- write(WM8731_REG_INTERFACE, 0x42); // I2S, 16 bit, MCLK master
- write(WM8731_REG_SAMPLING, 0x20); // 256*Fs, 44.1 kHz, MCLK/1
-
- // In order to prevent pops, the DAC should first be soft-muted (DACMU),
- // the output should then be de-selected from the line and headphone output
- // (DACSEL), then the DAC powered down (DACPD).
-
- write(WM8731_REG_DIGITAL, 0x08); // DAC soft mute
- write(WM8731_REG_ANALOG, 0x00); // disable all
-
- write(WM8731_REG_POWERDOWN, 0x00); // codec powerdown
-
- write(WM8731_REG_LHEADOUT, 0x80); // volume off
- write(WM8731_REG_RHEADOUT, 0x80);
-
- delay(100); // how long to power up?
-
- write(WM8731_REG_ACTIVE, 1);
- delay(5);
- write(WM8731_REG_DIGITAL, 0x00); // DAC unmuted
- write(WM8731_REG_ANALOG, 0x10); // DAC selected
-
- return true;
- }
-
- /******************************************************************/
-
- #define CHIP_ID 0x0000
- // 15:8 PARTID 0xA0 - 8 bit identifier for SGTL5000
- // 7:0 REVID 0x00 - revision number for SGTL5000.
-
- #define CHIP_DIG_POWER 0x0002
- // 6 ADC_POWERUP 1=Enable, 0=disable the ADC block, both digital & analog,
- // 5 DAC_POWERUP 1=Enable, 0=disable the DAC block, both analog and digital
- // 4 DAP_POWERUP 1=Enable, 0=disable the DAP block
- // 1 I2S_OUT_POWERUP 1=Enable, 0=disable the I2S data output
- // 0 I2S_IN_POWERUP 1=Enable, 0=disable the I2S data input
-
- #define CHIP_CLK_CTRL 0x0004
- // 5:4 RATE_MODE Sets the sample rate mode. MCLK_FREQ is still specified
- // relative to the rate in SYS_FS
- // 0x0 = SYS_FS specifies the rate
- // 0x1 = Rate is 1/2 of the SYS_FS rate
- // 0x2 = Rate is 1/4 of the SYS_FS rate
- // 0x3 = Rate is 1/6 of the SYS_FS rate
- // 3:2 SYS_FS Sets the internal system sample rate (default=2)
- // 0x0 = 32 kHz
- // 0x1 = 44.1 kHz
- // 0x2 = 48 kHz
- // 0x3 = 96 kHz
- // 1:0 MCLK_FREQ Identifies incoming SYS_MCLK frequency and if the PLL should be used
- // 0x0 = 256*Fs
- // 0x1 = 384*Fs
- // 0x2 = 512*Fs
- // 0x3 = Use PLL
- // The 0x3 (Use PLL) setting must be used if the SYS_MCLK is not
- // a standard multiple of Fs (256, 384, or 512). This setting can
- // also be used if SYS_MCLK is a standard multiple of Fs.
- // Before this field is set to 0x3 (Use PLL), the PLL must be
- // powered up by setting CHIP_ANA_POWER->PLL_POWERUP and
- // CHIP_ANA_POWER->VCOAMP_POWERUP. Also, the PLL dividers must
- // be calculated based on the external MCLK rate and
- // CHIP_PLL_CTRL register must be set (see CHIP_PLL_CTRL register
- // description details on how to calculate the divisors).
-
- #define CHIP_I2S_CTRL 0x0006
- // 8 SCLKFREQ Sets frequency of I2S_SCLK when in master mode (MS=1). When in slave
- // mode (MS=0), this field must be set appropriately to match SCLK input
- // rate.
- // 0x0 = 64Fs
- // 0x1 = 32Fs - Not supported for RJ mode (I2S_MODE = 1)
- // 7 MS Configures master or slave of I2S_LRCLK and I2S_SCLK.
- // 0x0 = Slave: I2S_LRCLK an I2S_SCLK are inputs
- // 0x1 = Master: I2S_LRCLK and I2S_SCLK are outputs
- // NOTE: If the PLL is used (CHIP_CLK_CTRL->MCLK_FREQ==0x3),
- // the SGTL5000 must be a master of the I2S port (MS==1)
- // 6 SCLK_INV Sets the edge that data (input and output) is clocked in on for I2S_SCLK
- // 0x0 = data is valid on rising edge of I2S_SCLK
- // 0x1 = data is valid on falling edge of I2S_SCLK
- // 5:4 DLEN I2S data length (default=1)
- // 0x0 = 32 bits (only valid when SCLKFREQ=0),
- // not valid for Right Justified Mode
- // 0x1 = 24 bits (only valid when SCLKFREQ=0)
- // 0x2 = 20 bits
- // 0x3 = 16 bits
- // 3:2 I2S_MODE Sets the mode for the I2S port
- // 0x0 = I2S mode or Left Justified (Use LRALIGN to select)
- // 0x1 = Right Justified Mode
- // 0x2 = PCM Format A/B
- // 0x3 = RESERVED
- // 1 LRALIGN I2S_LRCLK Alignment to data word. Not used for Right Justified mode
- // 0x0 = Data word starts 1 I2S_SCLK delay after I2S_LRCLK
- // transition (I2S format, PCM format A)
- // 0x1 = Data word starts after I2S_LRCLK transition (left
- // justified format, PCM format B)
- // 0 LRPOL I2S_LRCLK Polarity when data is presented.
- // 0x0 = I2S_LRCLK = 0 - Left, 1 - Right
- // 1x0 = I2S_LRCLK = 0 - Right, 1 - Left
- // The left subframe should be presented first regardless of
- // the setting of LRPOL.
-
- #define CHIP_SSS_CTRL 0x000A
- // 14 DAP_MIX_LRSWAP DAP Mixer Input Swap
- // 0x0 = Normal Operation
- // 0x1 = Left and Right channels for the DAP MIXER Input are swapped.
- // 13 DAP_LRSWAP DAP Mixer Input Swap
- // 0x0 = Normal Operation
- // 0x1 = Left and Right channels for the DAP Input are swapped
- // 12 DAC_LRSWAP DAC Input Swap
- // 0x0 = Normal Operation
- // 0x1 = Left and Right channels for the DAC are swapped
- // 10 I2S_LRSWAP I2S_DOUT Swap
- // 0x0 = Normal Operation
- // 0x1 = Left and Right channels for the I2S_DOUT are swapped
- // 9:8 DAP_MIX_SELECT Select data source for DAP mixer
- // 0x0 = ADC
- // 0x1 = I2S_IN
- // 0x2 = Reserved
- // 0x3 = Reserved
- // 7:6 DAP_SELECT Select data source for DAP
- // 0x0 = ADC
- // 0x1 = I2S_IN
- // 0x2 = Reserved
- // 0x3 = Reserved
- // 5:4 DAC_SELECT Select data source for DAC (default=1)
- // 0x0 = ADC
- // 0x1 = I2S_IN
- // 0x2 = Reserved
- // 0x3 = DAP
- // 1:0 I2S_SELECT Select data source for I2S_DOUT
- // 0x0 = ADC
- // 0x1 = I2S_IN
- // 0x2 = Reserved
- // 0x3 = DAP
-
- #define CHIP_ADCDAC_CTRL 0x000E
- // 13 VOL_BUSY_DAC_RIGHT Volume Busy DAC Right
- // 0x0 = Ready
- // 0x1 = Busy - This indicates the channel has not reached its
- // programmed volume/mute level
- // 12 VOL_BUSY_DAC_LEFT Volume Busy DAC Left
- // 0x0 = Ready
- // 0x1 = Busy - This indicates the channel has not reached its
- // programmed volume/mute level
- // 9 VOL_RAMP_EN Volume Ramp Enable (default=1)
- // 0x0 = Disables volume ramp. New volume settings take immediate
- // effect without a ramp
- // 0x1 = Enables volume ramp
- // This field affects DAC_VOL. The volume ramp effects both
- // volume settings and mute When set to 1 a soft mute is enabled.
- // 8 VOL_EXPO_RAMP Exponential Volume Ramp Enable
- // 0x0 = Linear ramp over top 4 volume octaves
- // 0x1 = Exponential ramp over full volume range
- // This bit only takes effect if VOL_RAMP_EN is 1.
- // 3 DAC_MUTE_RIGHT DAC Right Mute (default=1)
- // 0x0 = Unmute
- // 0x1 = Muted
- // If VOL_RAMP_EN = 1, this is a soft mute.
- // 2 DAC_MUTE_LEFT DAC Left Mute (default=1)
- // 0x0 = Unmute
- // 0x1 = Muted
- // If VOL_RAMP_EN = 1, this is a soft mute.
- // 1 ADC_HPF_FREEZE ADC High Pass Filter Freeze
- // 0x0 = Normal operation
- // 0x1 = Freeze the ADC high-pass filter offset register. The
- // offset continues to be subtracted from the ADC data stream.
- // 0 ADC_HPF_BYPASS ADC High Pass Filter Bypass
- // 0x0 = Normal operation
- // 0x1 = Bypassed and offset not updated
-
- #define CHIP_DAC_VOL 0x0010
- // 15:8 DAC_VOL_RIGHT DAC Right Channel Volume. Set the Right channel DAC volume
- // with 0.5017 dB steps from 0 to -90 dB
- // 0x3B and less = Reserved
- // 0x3C = 0 dB
- // 0x3D = -0.5 dB
- // 0xF0 = -90 dB
- // 0xFC and greater = Muted
- // If VOL_RAMP_EN = 1, there is an automatic ramp to the
- // new volume setting.
- // 7:0 DAC_VOL_LEFT DAC Left Channel Volume. Set the Left channel DAC volume
- // with 0.5017 dB steps from 0 to -90 dB
- // 0x3B and less = Reserved
- // 0x3C = 0 dB
- // 0x3D = -0.5 dB
- // 0xF0 = -90 dB
- // 0xFC and greater = Muted
- // If VOL_RAMP_EN = 1, there is an automatic ramp to the
- // new volume setting.
-
- #define CHIP_PAD_STRENGTH 0x0014
- // 9:8 I2S_LRCLK I2S LRCLK Pad Drive Strength (default=1)
- // Sets drive strength for output pads per the table below.
- // VDDIO 1.8 V 2.5 V 3.3 V
- // 0x0 = Disable
- // 0x1 = 1.66 mA 2.87 mA 4.02 mA
- // 0x2 = 3.33 mA 5.74 mA 8.03 mA
- // 0x3 = 4.99 mA 8.61 mA 12.05 mA
- // 7:6 I2S_SCLK I2S SCLK Pad Drive Strength (default=1)
- // 5:4 I2S_DOUT I2S DOUT Pad Drive Strength (default=1)
- // 3:2 CTRL_DATA I2C DATA Pad Drive Strength (default=3)
- // 1:0 CTRL_CLK I2C CLK Pad Drive Strength (default=3)
- // (all use same table as I2S_LRCLK)
-
- #define CHIP_ANA_ADC_CTRL 0x0020
- // 8 ADC_VOL_M6DB ADC Volume Range Reduction
- // This bit shifts both right and left analog ADC volume
- // range down by 6.0 dB.
- // 0x0 = No change in ADC range
- // 0x1 = ADC range reduced by 6.0 dB
- // 7:4 ADC_VOL_RIGHT ADC Right Channel Volume
- // Right channel analog ADC volume control in 1.5 dB steps.
- // 0x0 = 0 dB
- // 0x1 = +1.5 dB
- // ...
- // 0xF = +22.5 dB
- // This range is -6.0 dB to +16.5 dB if ADC_VOL_M6DB is set to 1.
- // 3:0 ADC_VOL_LEFT ADC Left Channel Volume
- // (same scale as ADC_VOL_RIGHT)
-
- #define CHIP_ANA_HP_CTRL 0x0022
- // 14:8 HP_VOL_RIGHT Headphone Right Channel Volume (default 0x18)
- // Right channel headphone volume control with 0.5 dB steps.
- // 0x00 = +12 dB
- // 0x01 = +11.5 dB
- // 0x18 = 0 dB
- // ...
- // 0x7F = -51.5 dB
- // 6:0 HP_VOL_LEFT Headphone Left Channel Volume (default 0x18)
- // (same scale as HP_VOL_RIGHT)
-
- #define CHIP_ANA_CTRL 0x0024
- // 8 MUTE_LO LINEOUT Mute, 0 = Unmute, 1 = Mute (default 1)
- // 6 SELECT_HP Select the headphone input, 0 = DAC, 1 = LINEIN
- // 5 EN_ZCD_HP Enable the headphone zero cross detector (ZCD)
- // 0x0 = HP ZCD disabled
- // 0x1 = HP ZCD enabled
- // 4 MUTE_HP Mute the headphone outputs, 0 = Unmute, 1 = Mute (default)
- // 2 SELECT_ADC Select the ADC input, 0 = Microphone, 1 = LINEIN
- // 1 EN_ZCD_ADC Enable the ADC analog zero cross detector (ZCD)
- // 0x0 = ADC ZCD disabled
- // 0x1 = ADC ZCD enabled
- // 0 MUTE_ADC Mute the ADC analog volume, 0 = Unmute, 1 = Mute (default)
-
- #define CHIP_LINREG_CTRL 0x0026
- // 6 VDDC_MAN_ASSN Determines chargepump source when VDDC_ASSN_OVRD is set.
- // 0x0 = VDDA
- // 0x1 = VDDIO
- // 5 VDDC_ASSN_OVRD Charge pump Source Assignment Override
- // 0x0 = Charge pump source is automatically assigned based
- // on higher of VDDA and VDDIO
- // 0x1 = the source of charge pump is manually assigned by
- // VDDC_MAN_ASSN If VDDIO and VDDA are both the same
- // and greater than 3.1 V, VDDC_ASSN_OVRD and
- // VDDC_MAN_ASSN should be used to manually assign
- // VDDIO as the source for charge pump.
- // 3:0 D_PROGRAMMING Sets the VDDD linear regulator output voltage in 50 mV steps.
- // Must clear the LINREG_SIMPLE_POWERUP and STARTUP_POWERUP bits
- // in the 0x0030 (CHIP_ANA_POWER) register after power-up, for
- // this setting to produce the proper VDDD voltage.
- // 0x0 = 1.60
- // 0xF = 0.85
-
- #define CHIP_REF_CTRL 0x0028 // bandgap reference bias voltage and currents
- // 8:4 VAG_VAL Analog Ground Voltage Control
- // These bits control the analog ground voltage in 25 mV steps.
- // This should usually be set to VDDA/2 or lower for best
- // performance (maximum output swing at minimum THD). This VAG
- // reference is also used for the DAC and ADC voltage reference.
- // So changing this voltage scales the output swing of the DAC
- // and the output signal of the ADC.
- // 0x00 = 0.800 V
- // 0x1F = 1.575 V
- // 3:1 BIAS_CTRL Bias control
- // These bits adjust the bias currents for all of the analog
- // blocks. By lowering the bias current a lower quiescent power
- // is achieved. It should be noted that this mode can affect
- // performance by 3-4 dB.
- // 0x0 = Nominal
- // 0x1-0x3=+12.5%
- // 0x4=-12.5%
- // 0x5=-25%
- // 0x6=-37.5%
- // 0x7=-50%
- // 0 SMALL_POP VAG Ramp Control
- // Setting this bit slows down the VAG ramp from ~200 to ~400 ms
- // to reduce the startup pop, but increases the turn on/off time.
- // 0x0 = Normal VAG ramp
- // 0x1 = Slow down VAG ramp
-
- #define CHIP_MIC_CTRL 0x002A // microphone gain & internal microphone bias
- // 9:8 BIAS_RESISTOR MIC Bias Output Impedance Adjustment
- // Controls an adjustable output impedance for the microphone bias.
- // If this is set to zero the micbias block is powered off and
- // the output is highZ.
- // 0x0 = Powered off
- // 0x1 = 2.0 kohm
- // 0x2 = 4.0 kohm
- // 0x3 = 8.0 kohm
- // 6:4 BIAS_VOLT MIC Bias Voltage Adjustment
- // Controls an adjustable bias voltage for the microphone bias
- // amp in 250 mV steps. This bias voltage setting should be no
- // more than VDDA-200 mV for adequate power supply rejection.
- // 0x0 = 1.25 V
- // ...
- // 0x7 = 3.00 V
- // 1:0 GAIN MIC Amplifier Gain
- // Sets the microphone amplifier gain. At 0 dB setting the THD
- // can be slightly higher than other paths- typically around
- // ~65 dB. At other gain settings the THD are better.
- // 0x0 = 0 dB
- // 0x1 = +20 dB
- // 0x2 = +30 dB
- // 0x3 = +40 dB
-
- #define CHIP_LINE_OUT_CTRL 0x002C
- // 11:8 OUT_CURRENT Controls the output bias current for the LINEOUT amplifiers. The
- // nominal recommended setting for a 10 kohm load with 1.0 nF load cap
- // is 0x3. There are only 5 valid settings.
- // 0x0=0.18 mA
- // 0x1=0.27 mA
- // 0x3=0.36 mA
- // 0x7=0.45 mA
- // 0xF=0.54 mA
- // 5:0 LO_VAGCNTRL LINEOUT Amplifier Analog Ground Voltage
- // Controls the analog ground voltage for the LINEOUT amplifiers
- // in 25 mV steps. This should usually be set to VDDIO/2.
- // 0x00 = 0.800 V
- // ...
- // 0x1F = 1.575 V
- // ...
- // 0x23 = 1.675 V
- // 0x24-0x3F are invalid
-
- #define CHIP_LINE_OUT_VOL 0x002E
- // 12:8 LO_VOL_RIGHT LINEOUT Right Channel Volume (default=4)
- // Controls the right channel LINEOUT volume in 0.5 dB steps.
- // Higher codes have more attenuation.
- // 4:0 LO_VOL_LEFT LINEOUT Left Channel Output Level (default=4)
- // Used to normalize the output level of the left line output
- // to full scale based on the values used to set
- // LINE_OUT_CTRL->LO_VAGCNTRL and CHIP_REF_CTRL->VAG_VAL.
- // In general this field should be set to:
- // 40*log((VAG_VAL)/(LO_VAGCNTRL)) + 15
- // Suggested values based on typical VDDIO and VDDA voltages.
- // VDDA VAG_VAL VDDIO LO_VAGCNTRL LO_VOL_*
- // 1.8 V 0.9 3.3 V 1.55 0x06
- // 1.8 V 0.9 1.8 V 0.9 0x0F
- // 3.3 V 1.55 1.8 V 0.9 0x19
- // 3.3 V 1.55 3.3 V 1.55 0x0F
- // After setting to the nominal voltage, this field can be used
- // to adjust the output level in +/-0.5 dB increments by using
- // values higher or lower than the nominal setting.
-
- #define CHIP_ANA_POWER 0x0030 // power down controls for the analog blocks.
- // The only other power-down controls are BIAS_RESISTOR in the MIC_CTRL register
- // and the EN_ZCD control bits in ANA_CTRL.
- // 14 DAC_MONO While DAC_POWERUP is set, this allows the DAC to be put into left only
- // mono operation for power savings. 0=mono, 1=stereo (default)
- // 13 LINREG_SIMPLE_POWERUP Power up the simple (low power) digital supply regulator.
- // After reset, this bit can be cleared IF VDDD is driven
- // externally OR the primary digital linreg is enabled with
- // LINREG_D_POWERUP
- // 12 STARTUP_POWERUP Power up the circuitry needed during the power up ramp and reset.
- // After reset this bit can be cleared if VDDD is coming from
- // an external source.
- // 11 VDDC_CHRGPMP_POWERUP Power up the VDDC charge pump block. If neither VDDA or VDDIO
- // is 3.0 V or larger this bit should be cleared before analog
- // blocks are powered up.
- // 10 PLL_POWERUP PLL Power Up, 0 = Power down, 1 = Power up
- // When cleared, the PLL is turned off. This must be set before
- // CHIP_CLK_CTRL->MCLK_FREQ is programmed to 0x3. The
- // CHIP_PLL_CTRL register must be configured correctly before
- // setting this bit.
- // 9 LINREG_D_POWERUP Power up the primary VDDD linear regulator, 0 = Power down, 1 = Power up
- // 8 VCOAMP_POWERUP Power up the PLL VCO amplifier, 0 = Power down, 1 = Power up
- // 7 VAG_POWERUP Power up the VAG reference buffer.
- // Setting this bit starts the power up ramp for the headphone
- // and LINEOUT. The headphone (and/or LINEOUT) powerup should
- // be set BEFORE clearing this bit. When this bit is cleared
- // the power-down ramp is started. The headphone (and/or LINEOUT)
- // powerup should stay set until the VAG is fully ramped down
- // (200 to 400 ms after clearing this bit).
- // 0x0 = Power down, 0x1 = Power up
- // 6 ADC_MONO While ADC_POWERUP is set, this allows the ADC to be put into left only
- // mono operation for power savings. This mode is useful when
- // only using the microphone input.
- // 0x0 = Mono (left only), 0x1 = Stereo
- // 5 REFTOP_POWERUP Power up the reference bias currents
- // 0x0 = Power down, 0x1 = Power up
- // This bit can be cleared when the part is a sleep state
- // to minimize analog power.
- // 4 HEADPHONE_POWERUP Power up the headphone amplifiers
- // 0x0 = Power down, 0x1 = Power up
- // 3 DAC_POWERUP Power up the DACs
- // 0x0 = Power down, 0x1 = Power up
- // 2 CAPLESS_HEADPHONE_POWERUP Power up the capless headphone mode
- // 0x0 = Power down, 0x1 = Power up
- // 1 ADC_POWERUP Power up the ADCs
- // 0x0 = Power down, 0x1 = Power up
- // 0 LINEOUT_POWERUP Power up the LINEOUT amplifiers
- // 0x0 = Power down, 0x1 = Power up
-
- #define CHIP_PLL_CTRL 0x0032
- // 15:11 INT_DIVISOR
- // 10:0 FRAC_DIVISOR
-
- #define CHIP_CLK_TOP_CTRL 0x0034
- // 11 ENABLE_INT_OSC Setting this bit enables an internal oscillator to be used for the
- // zero cross detectors, the short detect recovery, and the
- // charge pump. This allows the I2S clock to be shut off while
- // still operating an analog signal path. This bit can be kept
- // on when the I2S clock is enabled, but the I2S clock is more
- // accurate so it is preferred to clear this bit when I2S is present.
- // 3 INPUT_FREQ_DIV2 SYS_MCLK divider before PLL input
- // 0x0 = pass through
- // 0x1 = SYS_MCLK is divided by 2 before entering PLL
- // This must be set when the input clock is above 17 Mhz. This
- // has no effect when the PLL is powered down.
-
- #define CHIP_ANA_STATUS 0x0036
- // 9 LRSHORT_STS This bit is high whenever a short is detected on the left or right
- // channel headphone drivers.
- // 8 CSHORT_STS This bit is high whenever a short is detected on the capless headphone
- // common/center channel driver.
- // 4 PLL_IS_LOCKED This bit goes high after the PLL is locked.
-
- #define CHIP_ANA_TEST1 0x0038 // intended only for debug.
- #define CHIP_ANA_TEST2 0x003A // intended only for debug.
-
- #define CHIP_SHORT_CTRL 0x003C
- // 14:12 LVLADJR Right channel headphone short detector in 25 mA steps.
- // 0x3=25 mA
- // 0x2=50 mA
- // 0x1=75 mA
- // 0x0=100 mA
- // 0x4=125 mA
- // 0x5=150 mA
- // 0x6=175 mA
- // 0x7=200 mA
- // This trip point can vary by ~30% over process so leave plenty
- // of guard band to avoid false trips. This short detect trip
- // point is also effected by the bias current adjustments made
- // by CHIP_REF_CTRL->BIAS_CTRL and by CHIP_ANA_TEST1->HP_IALL_ADJ.
- // 10:8 LVLADJL Left channel headphone short detector in 25 mA steps.
- // (same scale as LVLADJR)
- // 6:4 LVLADJC Capless headphone center channel short detector in 50 mA steps.
- // 0x3=50 mA
- // 0x2=100 mA
- // 0x1=150 mA
- // 0x0=200 mA
- // 0x4=250 mA
- // 0x5=300 mA
- // 0x6=350 mA
- // 0x7=400 mA
- // 3:2 MODE_LR Behavior of left/right short detection
- // 0x0 = Disable short detector, reset short detect latch,
- // software view non-latched short signal
- // 0x1 = Enable short detector and reset the latch at timeout
- // (every ~50 ms)
- // 0x2 = This mode is not used/invalid
- // 0x3 = Enable short detector with only manual reset (have
- // to return to 0x0 to reset the latch)
- // 1:0 MODE_CM Behavior of capless headphone central short detection
- // (same settings as MODE_LR)
-
- #define DAP_CONTROL 0x0100
- #define DAP_PEQ 0x0102
- #define DAP_BASS_ENHANCE 0x0104
- #define DAP_BASS_ENHANCE_CTRL 0x0106
- #define DAP_AUDIO_EQ 0x0108
- #define DAP_SGTL_SURROUND 0x010A
- #define DAP_FILTER_COEF_ACCESS 0x010C
- #define DAP_COEF_WR_B0_MSB 0x010E
- #define DAP_COEF_WR_B0_LSB 0x0110
- #define DAP_AUDIO_EQ_BASS_BAND0 0x0116 // 115 Hz
- #define DAP_AUDIO_EQ_BAND1 0x0118 // 330 Hz
- #define DAP_AUDIO_EQ_BAND2 0x011A // 990 Hz
- #define DAP_AUDIO_EQ_BAND3 0x011C // 3000 Hz
- #define DAP_AUDIO_EQ_TREBLE_BAND4 0x011E // 9900 Hz
- #define DAP_MAIN_CHAN 0x0120
- #define DAP_MIX_CHAN 0x0122
- #define DAP_AVC_CTRL 0x0124
- #define DAP_AVC_THRESHOLD 0x0126
- #define DAP_AVC_ATTACK 0x0128
- #define DAP_AVC_DECAY 0x012A
- #define DAP_COEF_WR_B1_MSB 0x012C
- #define DAP_COEF_WR_B1_LSB 0x012E
- #define DAP_COEF_WR_B2_MSB 0x0130
- #define DAP_COEF_WR_B2_LSB 0x0132
- #define DAP_COEF_WR_A1_MSB 0x0134
- #define DAP_COEF_WR_A1_LSB 0x0136
- #define DAP_COEF_WR_A2_MSB 0x0138
- #define DAP_COEF_WR_A2_LSB 0x013A
-
- #define SGTL5000_I2C_ADDR 0x0A // CTRL_ADR0_CS pin low (normal configuration)
- //#define SGTL5000_I2C_ADDR 0x2A // CTRL_ADR0_CS pin high
-
-
-
- bool AudioControlSGTL5000::enable(void)
- {
- //unsigned int n;
-
- muted = true;
- Wire.begin();
- delay(5);
- //Serial.print("chip ID = ");
- //delay(5);
- //n = read(CHIP_ID);
- //Serial.println(n, HEX);
-
- write(CHIP_ANA_POWER, 0x4060); // VDDD is externally driven with 1.8V
- write(CHIP_LINREG_CTRL, 0x006C); // VDDA & VDDIO both over 3.1V
- write(CHIP_REF_CTRL, 0x01F1); // VAG=1.575 slow ramp, normal bias current
- write(CHIP_LINE_OUT_CTRL, 0x0322); // LO_VAGCNTRL=1.65V, OUT_CURRENT=0.36mA
- write(CHIP_SHORT_CTRL, 0x4446); // allow up to 125mA
- write(CHIP_ANA_CTRL, 0x0137); // enable zero cross detectors
- write(CHIP_ANA_POWER, 0x40FF); // power up: lineout, hp, adc, dac
- write(CHIP_DIG_POWER, 0x0073); // power up all digital stuff
- delay(400);
- // 40*log((1.575)/(1.65)) + 15 = 13.1391993746043 but it seems wrong, 5 is better...
- write(CHIP_LINE_OUT_VOL, 0x0505); // TODO: correct value for 3.3V
- write(CHIP_CLK_CTRL, 0x0004); // 44.1 kHz, 256*Fs
- write(CHIP_I2S_CTRL, 0x0130); // SCLK=32*Fs, 16bit, I2S format
- // default signal routing is ok?
- write(CHIP_SSS_CTRL, 0x0010); // ADC->I2S, I2S->DAC
- write(CHIP_ADCDAC_CTRL, 0x0000); // disable dac mute
- write(CHIP_DAC_VOL, 0x3C3C); // digital gain, 0dB
- write(CHIP_ANA_HP_CTRL, 0x7F7F); // set volume (lowest level)
- write(CHIP_ANA_CTRL, 0x0136); // enable zero cross detectors
- //mute = false;
- return true;
- }
-
- unsigned int AudioControlSGTL5000::read(unsigned int reg)
- {
- unsigned int val;
-
- Wire.beginTransmission(SGTL5000_I2C_ADDR);
- Wire.write(reg >> 8);
- Wire.write(reg);
- if (Wire.endTransmission(false) != 0) return 0;
- if (Wire.requestFrom(SGTL5000_I2C_ADDR, 2) < 2) return 0;
- val = Wire.read() << 8;
- val |= Wire.read();
- return val;
- }
-
- bool AudioControlSGTL5000::write(unsigned int reg, unsigned int val)
- {
- if (reg == CHIP_ANA_CTRL) ana_ctrl = val;
- Wire.beginTransmission(SGTL5000_I2C_ADDR);
- Wire.write(reg >> 8);
- Wire.write(reg);
- Wire.write(val >> 8);
- Wire.write(val);
- if (Wire.endTransmission() == 0) return true;
- return false;
- }
-
- unsigned int AudioControlSGTL5000::modify(unsigned int reg, unsigned int val, unsigned int iMask)
- {
- unsigned int val1 = (read(reg)&(~iMask))|val;
- if(!write(reg,val1)) return 0;
- return val1;
- }
-
- bool AudioControlSGTL5000::volumeInteger(unsigned int n)
- {
- if (n == 0) {
- muted = true;
- write(CHIP_ANA_HP_CTRL, 0x7F7F);
- return muteHeadphone();
- } else if (n > 0x80) {
- n = 0;
- } else {
- n = 0x80 - n;
- }
- if (muted) {
- muted = false;
- unmuteHeadphone();
- }
- n = n | (n << 8);
- return write(CHIP_ANA_HP_CTRL, n); // set volume
- }
-
- bool AudioControlSGTL5000::volume(float left, float right)
- {
- unsigned short m=((0x7F-calcVol(right,0x7F))<<8)|(0x7F-calcVol(left,0x7F));
- return write(CHIP_ANA_HP_CTRL, m);
- }
-
-
- // CHIP_LINE_OUT_VOL
- unsigned short AudioControlSGTL5000::lo_lvl(uint8_t n)
- {
- n&=31;
- return modify(CHIP_LINE_OUT_VOL,(n<<8)|n,(31<<8)|31);
- }
-
- unsigned short AudioControlSGTL5000::lo_lvl(uint8_t left, uint8_t right)
- {
- left&=31;
- right&=31;
- return modify(CHIP_LINE_OUT_VOL,(right<<8)|left,(31<<8)|31);
- }
-
- unsigned short AudioControlSGTL5000::dac_vol(float n) // set both directly
- {
- if(read(CHIP_ADCDAC_CTRL)&(3<<2)!=((n>0 ? 0:3)<<2)) modify(CHIP_ADCDAC_CTRL,(n>0 ? 0:3)<<2,3<<2);
- unsigned char m=calcVol(n,0xC0);
- return modify(CHIP_DAC_VOL,((0xFC-m)<<8)|(0xFC-m),65535);
- }
- unsigned short AudioControlSGTL5000::dac_vol(float left, float right)
- {
- unsigned short adcdac=((right>0 ? 0:2)|(left>0 ? 0:1))<<2;
- if(read(CHIP_ADCDAC_CTRL)&(3<<2)!=adcdac) modify(CHIP_ADCDAC_CTRL,adcdac,1<<2);
- unsigned short m=(0xFC-calcVol(right,0xC0))<<8|(0xFC-calcVol(left,0xC0));
- return modify(CHIP_DAC_VOL,m,65535);
- }
- // DAP_CONTROL
- unsigned short AudioControlSGTL5000::dap_mix_enable(uint8_t n)
- {
- return modify(DAP_CONTROL,(n&1)<<4,1<<4);
- }
- unsigned short AudioControlSGTL5000::dap_enable(uint8_t n)
- {
- if(n) n=1;
- unsigned char DAC=1+(2*n); // I2S_IN if n==0 else DAP
- modify(DAP_CONTROL,n,1);
- return modify(CHIP_SSS_CTRL,(0<<6)|(DAC<<4),(3<<6)|(3<<4));
- }
-
- unsigned short AudioControlSGTL5000::dap_enable(void)
- {
- return dap_enable(1);
- }
-
- // DAP_PEQ
- unsigned short AudioControlSGTL5000::dap_peqs(uint8_t n) // valid to n&7, 0 thru 7 filters enabled.
- {
- return modify(DAP_PEQ,(n&7),7);
- }
-
- // DAP_AUDIO_EQ
- unsigned short AudioControlSGTL5000::dap_audio_eq(uint8_t n) // 0=NONE, 1=PEQ (7 IIR Biquad filters), 2=TONE (tone), 3=GEQ (5 band EQ)
- {
- return modify(DAP_AUDIO_EQ,n&3,3);
- }
-
-
- /******************************************************************/
-
- void AudioFilterFIR::begin(short *cp,int n_coeffs)
- {
- // pointer to coefficients
- coeff_p = cp;
- // Initialize FIR instances for the left and right channels
- if(coeff_p && (coeff_p != FIR_PASSTHRU)) {
- arm_fir_init_q15(&l_fir_inst, n_coeffs, coeff_p, &l_StateQ15[0], AUDIO_BLOCK_SAMPLES);
- arm_fir_init_q15(&r_fir_inst, n_coeffs, coeff_p, &r_StateQ15[0], AUDIO_BLOCK_SAMPLES);
- }
- }
-
- // This has the same effect as begin(NULL,0);
- void AudioFilterFIR::stop(void)
- {
- coeff_p = NULL;
- }
-
-
- void AudioFilterFIR::update(void)
- {
- audio_block_t *block,*b_new;
-
- // If there's no coefficient table, give up.
- if(coeff_p == NULL)return;
-
- // do passthru
- if(coeff_p == FIR_PASSTHRU) {
- // Just passthrough
- block = receiveWritable(0);
- if(block) {
- transmit(block,0);
- release(block);
- }
- block = receiveWritable(1);
- if(block) {
- transmit(block,1);
- release(block);
- }
- return;
- }
- // Left Channel
- block = receiveWritable(0);
- // get a block for the FIR output
- b_new = allocate();
- if(block && b_new) {
- arm_fir_q15(&l_fir_inst, (q15_t *)block->data, (q15_t *)b_new->data, AUDIO_BLOCK_SAMPLES);
- // send the FIR output to the left channel
- transmit(b_new,0);
- }
- if(block)release(block);
- if(b_new)release(b_new);
-
- // Right Channel
- block = receiveWritable(1);
- b_new = allocate();
- if(block && b_new) {
- arm_fir_q15(&r_fir_inst, (q15_t *)block->data, (q15_t *)b_new->data, AUDIO_BLOCK_SAMPLES);
- transmit(b_new,1);
- }
- if(block)release(block);
- if(b_new)release(b_new);
- }
-
-
- /******************************************************************/
- // A u d i o E f f e c t F l a n g e
- // Written by Pete (El Supremo) Jan 2014
- // 140207 - fix calculation of delay_rate_incr which is expressed as
- // a fraction of 2*PI
- // 140207 - cosmetic fix to begin()
-
- // circular addressing indices for left and right channels
- short AudioEffectFlange::l_circ_idx;
- short AudioEffectFlange::r_circ_idx;
-
- short * AudioEffectFlange::l_delayline = NULL;
- short * AudioEffectFlange::r_delayline = NULL;
-
- // User-supplied offset for the delayed sample
- // but start with passthru
- int AudioEffectFlange::delay_offset_idx = DELAY_PASSTHRU;
- int AudioEffectFlange::delay_length;
-
- int AudioEffectFlange::delay_depth;
- int AudioEffectFlange::delay_rate_incr;
- unsigned int AudioEffectFlange::l_delay_rate_index;
- unsigned int AudioEffectFlange::r_delay_rate_index;
- // fails if the user provides unreasonable values but will
- // coerce them and go ahead anyway. e.g. if the delay offset
- // is >= CHORUS_DELAY_LENGTH, the code will force it to
- // CHORUS_DELAY_LENGTH-1 and return false.
- // delay_rate is the rate (in Hz) of the sine wave modulation
- // delay_depth is the maximum variation around delay_offset
- // i.e. the total offset is delay_offset + delay_depth * sin(delay_rate)
- boolean AudioEffectFlange::begin(short *delayline,int d_length,int delay_offset,int d_depth,float delay_rate)
- {
- boolean all_ok = true;
-
- if(0) {
- Serial.print("AudioEffectFlange.begin(offset = ");
- Serial.print(delay_offset);
- Serial.print(", depth = ");
- Serial.print(d_depth);
- Serial.print(", rate = ");
- Serial.print(delay_rate,3);
- Serial.println(")");
- Serial.print(" FLANGE_DELAY_LENGTH = ");
- Serial.println(d_length);
- }
- delay_length = d_length/2;
- l_delayline = delayline;
- r_delayline = delayline + delay_length;
-
- delay_depth = d_depth;
- // initial index
- l_delay_rate_index = 0;
- r_delay_rate_index = 0;
- l_circ_idx = 0;
- r_circ_idx = 0;
- delay_rate_incr = delay_rate/44100.*2147483648.;
- //Serial.println(delay_rate_incr,HEX);
-
- delay_offset_idx = delay_offset;
- // Allow the passthru code to go through
- if(delay_offset_idx < -1) {
- delay_offset_idx = 0;
- all_ok = false;
- }
- if(delay_offset_idx >= delay_length) {
- delay_offset_idx = delay_length - 1;
- all_ok = false;
- }
- return(all_ok);
- }
-
-
- boolean AudioEffectFlange::modify(int delay_offset,int d_depth,float delay_rate)
- {
- boolean all_ok = true;
-
- delay_depth = d_depth;
-
- delay_rate_incr = delay_rate/44100.*2147483648.;
-
- delay_offset_idx = delay_offset;
- // Allow the passthru code to go through
- if(delay_offset_idx < -1) {
- delay_offset_idx = 0;
- all_ok = false;
- }
- if(delay_offset_idx >= delay_length) {
- delay_offset_idx = delay_length - 1;
- all_ok = false;
- }
- l_delay_rate_index = 0;
- r_delay_rate_index = 0;
- l_circ_idx = 0;
- r_circ_idx = 0;
- return(all_ok);
- }
-
- void AudioEffectFlange::update(void)
- {
- audio_block_t *block;
- int idx;
- short *bp;
- short frac;
- int idx1;
-
- if(l_delayline == NULL)return;
- if(r_delayline == NULL)return;
-
- // do passthru
- if(delay_offset_idx == DELAY_PASSTHRU) {
- // Just passthrough
- block = receiveWritable(0);
- if(block) {
- bp = block->data;
- for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
- l_circ_idx++;
- if(l_circ_idx >= delay_length) {
- l_circ_idx = 0;
- }
- l_delayline[l_circ_idx] = *bp++;
- }
- transmit(block,0);
- release(block);
- }
- block = receiveWritable(1);
- if(block) {
- bp = block->data;
- for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
- r_circ_idx++;
- if(r_circ_idx >= delay_length) {
- r_circ_idx = 0;
- }
- r_delayline[r_circ_idx] = *bp++;
- }
- transmit(block,1);
- release(block);
- }
- return;
- }
-
- // L E F T C H A N N E L
-
- block = receiveWritable(0);
- if(block) {
- bp = block->data;
- for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
- l_circ_idx++;
- if(l_circ_idx >= delay_length) {
- l_circ_idx = 0;
- }
- l_delayline[l_circ_idx] = *bp;
- idx = arm_sin_q15( (q15_t)((l_delay_rate_index >> 16) & 0x7fff));
- idx = (idx * delay_depth) >> 15;
- //Serial.println(idx);
- idx = l_circ_idx - (delay_offset_idx + idx);
- if(idx < 0) {
- idx += delay_length;
- }
- if(idx >= delay_length) {
- idx -= delay_length;
- }
-
- if(frac < 0)
- idx1 = idx - 1;
- else
- idx1 = idx + 1;
- if(idx1 < 0) {
- idx1 += delay_length;
- }
- if(idx1 >= delay_length) {
- idx1 -= delay_length;
- }
- frac = (l_delay_rate_index >> 1) &0x7fff;
- frac = (( (int)(l_delayline[idx1] - l_delayline[idx])*frac) >> 15);
-
- *bp++ = (l_delayline[l_circ_idx]
- + l_delayline[idx] + frac
- )/2;
-
- l_delay_rate_index += delay_rate_incr;
- if(l_delay_rate_index & 0x80000000) {
- l_delay_rate_index &= 0x7fffffff;
- }
- }
- // send the effect output to the left channel
- transmit(block,0);
- release(block);
- }
-
- // R I G H T C H A N N E L
-
- block = receiveWritable(1);
- if(block) {
- bp = block->data;
- for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
- r_circ_idx++;
- if(r_circ_idx >= delay_length) {
- r_circ_idx = 0;
- }
- r_delayline[r_circ_idx] = *bp;
- idx = arm_sin_q15( (q15_t)((r_delay_rate_index >> 16)&0x7fff));
- idx = (idx * delay_depth) >> 15;
-
- idx = r_circ_idx - (delay_offset_idx + idx);
- if(idx < 0) {
- idx += delay_length;
- }
- if(idx >= delay_length) {
- idx -= delay_length;
- }
-
- if(frac < 0)
- idx1 = idx - 1;
- else
- idx1 = idx + 1;
- if(idx1 < 0) {
- idx1 += delay_length;
- }
- if(idx1 >= delay_length) {
- idx1 -= delay_length;
- }
- frac = (r_delay_rate_index >> 1) &0x7fff;
- frac = (( (int)(r_delayline[idx1] - r_delayline[idx])*frac) >> 15);
-
- *bp++ = (r_delayline[r_circ_idx]
- + r_delayline[idx] + frac
- )/2;
-
- r_delay_rate_index += delay_rate_incr;
- if(r_delay_rate_index & 0x80000000) {
- r_delay_rate_index &= 0x7fffffff;
- }
-
- }
- // send the effect output to the right channel
- transmit(block,1);
- release(block);
- }
- }
-
-
-
- /******************************************************************/
-
- // A u d i o E f f e c t C h o r u s
- // Written by Pete (El Supremo) Jan 2014
-
- // circular addressing indices for left and right channels
- short AudioEffectChorus::l_circ_idx;
- short AudioEffectChorus::r_circ_idx;
-
- short * AudioEffectChorus::l_delayline = NULL;
- short * AudioEffectChorus::r_delayline = NULL;
- int AudioEffectChorus::delay_length;
- // An initial value of zero indicates passthru
- int AudioEffectChorus::num_chorus = 0;
-
-
- // All three must be valid.
- boolean AudioEffectChorus::begin(short *delayline,int d_length,int n_chorus)
- {
- Serial.print("AudioEffectChorus.begin(Chorus delay line length = ");
- Serial.print(d_length);
- Serial.print(", n_chorus = ");
- Serial.print(n_chorus);
- Serial.println(")");
-
- l_delayline = NULL;
- r_delayline = NULL;
- delay_length = 0;
- l_circ_idx = 0;
- r_circ_idx = 0;
-
- if(delayline == NULL) {
- return(false);
- }
- if(d_length < 10) {
- return(false);
- }
- if(n_chorus < 1) {
- return(false);
- }
-
- l_delayline = delayline;
- r_delayline = delayline + d_length/2;
- delay_length = d_length/2;
- num_chorus = n_chorus;
-
- return(true);
- }
-
- // This has the same effect as begin(NULL,0);
- void AudioEffectChorus::stop(void)
- {
-
- }
-
- void AudioEffectChorus::modify(int n_chorus)
- {
- num_chorus = n_chorus;
- }
-
- int iabs(int x)
- {
- if(x < 0)return(-x);
- return(x);
- }
- //static int d_count = 0;
-
- int last_idx = 0;
- void AudioEffectChorus::update(void)
- {
- audio_block_t *block;
- short *bp;
- int sum;
- int c_idx;
-
- if(l_delayline == NULL)return;
- if(r_delayline == NULL)return;
-
- // do passthru
- // It stores the unmodified data in the delay line so that
- // it isn't as likely to click
- if(num_chorus < 1) {
- // Just passthrough
- block = receiveWritable(0);
- if(block) {
- bp = block->data;
- for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
- l_circ_idx++;
- if(l_circ_idx >= delay_length) {
- l_circ_idx = 0;
- }
- l_delayline[l_circ_idx] = *bp++;
- }
- transmit(block,0);
- release(block);
- }
- block = receiveWritable(1);
- if(block) {
- bp = block->data;
- for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
- r_circ_idx++;
- if(r_circ_idx >= delay_length) {
- r_circ_idx = 0;
- }
- r_delayline[r_circ_idx] = *bp++;
- }
- transmit(block,1);
- release(block);
- }
- return;
- }
-
- // L E F T C H A N N E L
-
- block = receiveWritable(0);
- if(block) {
- bp = block->data;
- for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
- l_circ_idx++;
- if(l_circ_idx >= delay_length) {
- l_circ_idx = 0;
- }
- l_delayline[l_circ_idx] = *bp;
- sum = 0;
- c_idx = l_circ_idx;
- for(int k = 0; k < num_chorus; k++) {
- sum += l_delayline[c_idx];
- if(num_chorus > 1)c_idx -= delay_length/(num_chorus - 1) - 1;
- if(c_idx < 0) {
- c_idx += delay_length;
- }
- }
- *bp++ = sum/num_chorus;
- }
-
- // send the effect output to the left channel
- transmit(block,0);
- release(block);
- }
-
- // R I G H T C H A N N E L
-
- block = receiveWritable(1);
- if(block) {
- bp = block->data;
- for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
- r_circ_idx++;
- if(r_circ_idx >= delay_length) {
- r_circ_idx = 0;
- }
- r_delayline[r_circ_idx] = *bp;
- sum = 0;
- c_idx = r_circ_idx;
- for(int k = 0; k < num_chorus; k++) {
- sum += r_delayline[c_idx];
- if(num_chorus > 1)c_idx -= delay_length/(num_chorus - 1) - 1;
- if(c_idx < 0) {
- c_idx += delay_length;
- }
- }
- *bp++ = sum/num_chorus;
- }
-
- // send the effect output to the left channel
- transmit(block,1);
- release(block);
- }
- }
-
-
- // DAP_AUDIO_EQ_BASS_BAND0 & DAP_AUDIO_EQ_BAND1 & DAP_AUDIO_EQ_BAND2 etc etc
- unsigned short AudioControlSGTL5000::dap_audio_eq_band(uint8_t bandNum, float n) // by signed percentage -100/+100; dap_audio_eq(3);
- { // 0x00==-12dB, 0x2F==0dB, 0x5F==12dB
- n=((n/100)*48)+0.499;
- if(n<-47) n=-47;
- if(n>48) n=48;
- n+=47;
- return modify(DAP_AUDIO_EQ_BASS_BAND0+(bandNum*2),(unsigned int)n,127);
- }
- void AudioControlSGTL5000::dap_audio_eq_geq(float bass, float mid_bass, float midrange, float mid_treble, float treble)
- {
- dap_audio_eq_band(0,bass);
- dap_audio_eq_band(1,mid_bass);
- dap_audio_eq_band(2,midrange);
- dap_audio_eq_band(3,mid_treble);
- dap_audio_eq_band(4,treble);
- }
- void AudioControlSGTL5000::dap_audio_eq_tone(float bass, float treble) // dap_audio_eq(2);
- {
- dap_audio_eq_band(0,bass);
- dap_audio_eq_band(4,treble);
- }
-
- // SGTL5000 PEQ Coefficient loader
- void AudioControlSGTL5000::load_peq(uint8_t filterNum, int *filterParameters)
- {
- // 1111 11111111 11111111
-
- write(DAP_COEF_WR_B0_MSB,(*filterParameters>>4)&65535);
- write(DAP_COEF_WR_B0_LSB,(*filterParameters++)&15);
- write(DAP_COEF_WR_B1_MSB,(*filterParameters>>4)&65535);
- write(DAP_COEF_WR_B1_LSB,(*filterParameters++)&15);
- write(DAP_COEF_WR_B2_MSB,(*filterParameters>>4)&65535);
- write(DAP_COEF_WR_B2_LSB,(*filterParameters++)&15);
- write(DAP_COEF_WR_A1_MSB,(*filterParameters>>4)&65535);
- write(DAP_COEF_WR_A1_LSB,(*filterParameters++)&15);
- write(DAP_COEF_WR_A2_MSB,(*filterParameters>>4)&65535);
- write(DAP_COEF_WR_A2_LSB,(*filterParameters++)&15);
- write(DAP_FILTER_COEF_ACCESS,(uint16_t)0x100|filterNum);
- delay(10); // seems necessary, didn't work for 1ms.
- modify(DAP_FILTER_COEF_ACCESS,(uint16_t)filterNum,15);
- }
-
- unsigned char AudioControlSGTL5000::calcVol(float n, unsigned char range)
- {
- n=(n*(((float)range)/100))+0.499;
- if ((unsigned char)n>range) n=range;
- return (unsigned char)n;
- }
-
- // if(SGTL5000_PEQ) quantization_unit=524288; if(AudioFilterBiquad) quantization_unit=2147483648;
- void calcBiquad(uint8_t filtertype, float fC, float dB_Gain, float Q, uint32_t quantization_unit, uint32_t fS, int *coef)
- {
-
- // I used resources like http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
- // to make this routine, I tested most of the filter types and they worked. Such filters have limits and
- // before calling this routine with varying values the end user should check that those values are limited
- // to valid results.
-
- float A;
- if(filtertype<FILTER_PARAEQ) A=pow(10,dB_Gain/20); else A=pow(10,dB_Gain/40);
- float W0 = 2*3.14159265358979323846*fC/fS;
- float cosw=cos(W0);
- float sinw=sin(W0);
- //float alpha = sinw*sinh((log(2)/2)*BW*W0/sinw);
- //float beta = sqrt(2*A);
- float alpha = sinw / (2 * Q);
- float beta = sqrt(A)/Q;
- float b0,b1,b2,a0,a1,a2;
-
- switch(filtertype) {
- case FILTER_LOPASS:
- b0 = (1.0F - cosw) * 0.5F; // =(1-COS($H$2))/2
- b1 = 1.0F - cosw;
- b2 = (1.0F - cosw) * 0.5F;
- a0 = 1.0F + alpha;
- a1 = 2.0F * cosw;
- a2 = alpha - 1.0F;
- break;
- case FILTER_HIPASS:
- b0 = (1.0F + cosw) * 0.5F;
- b1 = -(cosw + 1.0F);
- b2 = (1.0F + cosw) * 0.5F;
- a0 = 1.0F + alpha;
- a1 = 2.0F * cosw;
- a2 = alpha - 1.0F;
- break;
- case FILTER_BANDPASS:
- b0 = alpha;
- b1 = 0.0F;
- b2 = -alpha;
- a0 = 1.0F + alpha;
- a1 = 2.0F * cosw;
- a2 = alpha - 1.0F;
- break;
- case FILTER_NOTCH:
- b0=1;
- b1=-2*cosw;
- b2=1;
- a0=1+alpha;
- a1=2*cosw;
- a2=-(1-alpha);
- break;
- case FILTER_PARAEQ:
- b0 = 1 + (alpha*A);
- b1 =-2 * cosw;
- b2 = 1 - (alpha*A);
- a0 = 1 + (alpha/A);
- a1 = 2 * cosw;
- a2 =-(1-(alpha/A));
- break;
- case FILTER_LOSHELF:
- b0 = A * ((A+1.0F) - ((A-1.0F)*cosw) + (beta*sinw));
- b1 = 2.0F * A * ((A-1.0F) - ((A+1.0F)*cosw));
- b2 = A * ((A+1.0F) - ((A-1.0F)*cosw) - (beta*sinw));
- a0 = (A+1.0F) + ((A-1.0F)*cosw) + (beta*sinw);
- a1 = 2.0F * ((A-1.0F) + ((A+1.0F)*cosw));
- a2 = -((A+1.0F) + ((A-1.0F)*cosw) - (beta*sinw));
- break;
- case FILTER_HISHELF:
- b0 = A * ((A+1.0F) + ((A-1.0F)*cosw) + (beta*sinw));
- b1 = -2.0F * A * ((A-1.0F) + ((A+1.0F)*cosw));
- b2 = A * ((A+1.0F) + ((A-1.0F)*cosw) - (beta*sinw));
- a0 = (A+1.0F) - ((A-1.0F)*cosw) + (beta*sinw);
- a1 = -2.0F * ((A-1.0F) - ((A+1.0F)*cosw));
- a2 = -((A+1.0F) - ((A-1.0F)*cosw) - (beta*sinw));
- }
-
- a0=(a0*2)/(float)quantization_unit; // once here instead of five times there...
- b0/=a0;
- *coef++=(int)(b0+0.499);
- b1/=a0;
- *coef++=(int)(b1+0.499);
- b2/=a0;
- *coef++=(int)(b2+0.499);
- a1/=a0;
- *coef++=(int)(a1+0.499);
- a2/=a0;
- *coef++=(int)(a2+0.499);
- }
-
- /******************************************************************/
-
- // A u d i o T o n e S w e e p
- // Written by Pete (El Supremo) Feb 2014
-
-
- boolean AudioToneSweep::begin(short t_amp,int t_lo,int t_hi,float t_time)
- {
- double tone_tmp;
-
- if(0) {
- Serial.print("AudioToneSweep.begin(tone_amp = ");
- Serial.print(t_amp);
- Serial.print(", tone_lo = ");
- Serial.print(t_lo);
- Serial.print(", tone_hi = ");
- Serial.print(t_hi);
- Serial.print(", tone_time = ");
- Serial.print(t_time,1);
- Serial.println(")");
- }
- tone_amp = 0;
- if(t_amp < 0)return false;
- if(t_lo < 1)return false;
- if(t_hi < 1)return false;
- if(t_hi >= 44100/2)return false;
- if(t_lo >= 44100/2)return false;
- if(t_time < 0)return false;
- tone_lo = t_lo;
- tone_hi = t_hi;
- tone_phase = 0;
-
- tone_amp = t_amp;
- // Limit the output amplitude to prevent aliasing
- // until I can figure out why this "overtops"
- // above 29000.
- if(tone_amp > 29000)tone_amp = 29000;
- tone_tmp = tone_hi - tone_lo;
- tone_sign = 1;
- tone_freq = tone_lo*0x100000000LL;
- if(tone_tmp < 0) {
- tone_sign = -1;
- tone_tmp = -tone_tmp;
- }
- tone_tmp = tone_tmp/t_time/44100.;
- tone_incr = (tone_tmp * 0x100000000LL);
- sweep_busy = 1;
- return(true);
- }
-
-
-
- unsigned char AudioToneSweep::busy(void)
- {
- return(sweep_busy);
- }
-
- int b_count = 0;
- void AudioToneSweep::update(void)
- {
- audio_block_t *block;
- short *bp;
- int i;
-
- if(!sweep_busy)return;
-
- // L E F T C H A N N E L O N L Y
- block = allocate();
- if(block) {
- bp = block->data;
- // Generate the sweep
- for(i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
- *bp++ = (short)(( (short)(arm_sin_q31((uint32_t)((tone_phase >> 15)&0x7fffffff))>>16) *tone_amp) >> 16);
- uint64_t tone_tmp = (0x400000000000LL * (int)((tone_freq >> 32)&0x7fffffff))/44100;
-
- tone_phase += tone_tmp;
- if(tone_phase & 0x800000000000LL)tone_phase &= 0x7fffffffffffLL;
-
- if(tone_sign > 0) {
- if((tone_freq >> 32) > tone_hi) {
- sweep_busy = 0;
- break;
- }
- tone_freq += tone_incr;
- } else {
- if((tone_freq >> 32) < tone_hi) {
- sweep_busy = 0;
-
- break;
- }
- tone_freq -= tone_incr;
- }
- }
- while(i < AUDIO_BLOCK_SAMPLES) {
- *bp++ = 0;
- i++;
- }
- b_count++;
- // send the samples to the left channel
- transmit(block,0);
- release(block);
- }
- }
|